Digital signal processing using variable sampling and its applications

可变采样数字信号处理及其应用

• 批准号: 12650379
• 负责人: TADOKORO Yoshiaki
• 金额: $ 1.86万
• 依托单位: Toyohashi University of Technology
• 依托单位国家: 日本
• 项目类别: Grant-in-Aid for Scientific Research (C)
• 财政年份: 2000
• 资助国家: 日本
• 起止时间: 2000 至 2001
• 项目状态: 已结题
• 来源: https://kaken.nii.ac.jp/grant/KAKENHI-PROJECT-12650379/
• 关键词: 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

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

已经进行了这项研究以开发新的数字信号处理注意到信号时间，即在采样时？，多少延迟？以及何时输出？结果，我们可以使用添加和减法来开发以下新的数字信号处理算法及其应用。1。使用自适应采样率（1）使用同步添加和减去累加值的频率估计算法（1）频率估计算法通过同步添加和减法，我们可以估算输入频率并基于估计频率设置新的采样率（（ 2）使用自适应缺口过滤器的频率估计算法，我们更改自适应置换过滤器的延迟时间，以降低自适应凹口滤波器的振幅或根据自适应凹口滤波器和零越过点的周期估计输入频率2。离散的傅立叶变换算法使用多胎采样（1）多段离散傅里叶变换算法（MR-DFT）在此MR-DFT中，每个信号分量的极端值都是采样的，并使用添加和减法来累积。该算法是通过一个傅立叶系数的一个乘法来计算的。（2）梳子傅立叶变换（CFT）该算法可以应用于由任意信号频率组成的信号，并通过与O（n）的乘数数量进行计算，与O（n）的乘法数量进行计算。 （n^3）在信号组件数为n.3时同时方程方法中。转录系统（1）转录系统，使用基于级联的梳子过滤器的同步加法和减法（2）转录系统，以及自适应梳子过滤器（3）基于梳子滤波器输出的单数值分解（SVD）基于平行梳子过滤器的转录系统，或相邻梳子滤波器输出之间的相似性

项目成果

Yoshiaki Tadokoro: "Musical pitch and instrument estimation of polyphony using comb filters for transcription"The 4 th World Multi-Conference on Circuits, Systems, Communications and Computers. 315-319 (2000)

Yoshiaki Tadokoro：“使用梳状滤波器进行转录的复调音调和乐器估计”第四届世界电路、系统、通信和计算机会议。

Yoshiaki Tadokoro: "Comb Fourier transform with fewer multiplications"Eurrasip Conference on Digital Signal Processing for Multimedia Communications and Services (ECMCS) ECMCS-154. (CD-ROM). 154(4p) (2001)

Yoshiaki Tadokoro：“用更少的乘法进行梳状傅里叶变换”Eurrasip 多媒体通信和服务数字信号处理会议 (ECMCS) ECMCS-154。

工藤憲昌: "白色雑音と干渉がある場合の信号成分のフーリエ係数推定法"電子情報通信学会論文誌. J83-A・12. 1467-1476 (2000)

Norimasa Kudo：“存在白噪声和干扰的情况下信号分量的傅里叶系数估计方法”电子信息通信工程师学会学报J83-A·12（2000）。

Yoshiaki Tadokoro: "Comb Fourier transform with fewer multiplications"Europsip Conference on Digital Signal Processing for Multimedia Communications and Services (ECMCS). ECMCS-154(CD). 154 (2001)

Yoshiaki Tadokoro：“用更少的乘法进行梳状傅立叶变换”欧洲多媒体通信和服务数字信号处理 (ECMCS) 会议。

Yoshiaki Tadokoro: "New adaptive frequency estimation methods using variable sampling processing"IEEE Adaptive Systems for Signal Processing Communications, and Control Symposium (AS-SPCC). 241-244 (2000)

Yoshiaki Tadokoro：“使用可变采样处理的新型自适应频率估计方法”IEEE 信号处理通信自适应系统和控制研讨会（AS-SPCC）。