時間頻率濾波 的英文怎麼說

中文拼音 [shíjiānbīnlǜlǜbō]
時間頻率濾波 英文
time frequency filtering
  • 時 : shí］Ⅰ名1 （比較長的一段時間）time; times; days:當時at that time; in those days; 古時 ancient tim...
  • 間 : 間Ⅰ名詞1 （中間） between; among 2 （一定的空間或時間里） with a definite time or space 3 （一間...
  • 頻 : Ⅰ形容詞（次數多） frequent Ⅱ副詞（屢次） frequently; repeatedly Ⅲ名詞1 [物理學] （物體每秒鐘振動...
  • 率 : 率名詞（比值） rate; ratio; proportion
  • 濾 : 動詞（除去液體雜質） filter; strain
  • 波 : Ⅰ名詞1 （波浪） wave 2 [物理學] （振動傳播的過程） wave 3 （意外變化） an unexpected turn of even...
  • 時間 : time; hour; 北京時間十九點整19 hours beijing time; 上課時間school hours; 時間與空間 time and spac...
  • 頻率 : frequency; rate
  1. In this paper, the concept of acceleration ambiguity function ( aaf ) and acceleration resolution is put forward for the first time. using aaf, the effects of target ' s relative acceleration on several outputs of a linear - phase matched filter are analyzed, such as the output signal - noise - ratio loss, the doppler resolution, the constraint on optimal accumulative time ( opt ) and their tolerable limits

    論文首次提出並研究了加速度模糊函數和加速度分辨力的有關概念，並以加速度模糊函數為分析工具，詳細討論了加速度對線性相位匹配濾波器的輸出信噪比的損失程度、對多卜勒頻率分辨能力的影響程度、對最優相參積累時間的約束關系以及線性相位匹配濾波器輸出受加速度影響的容限等問題。
  2. Section ii describes the design approach and implementation of speech module on mcf5249 coldfire core. the speech codec optimizes g. 729a codes and added voice activity detection of g. 729b to save bandwidth ; the implementation of acoustic echo cancellation uses nlms algorithm and it can reduce echo though designing adaptive fir filter and speech detector ; the dtmf and cpt generate signal using two second order digital sinusoidal oscillators and detect signal by picking up the frequency information. but only get the frequency information is not enough in cpt detector, this thesis introduces a method

    其中對語音編解碼器的設計採用優化g . 729a代碼達到設計要求，並在此基礎上加入g . 729b的靜音檢測模塊，以進一步降低網路傳輸帶寬;對回聲消除器的設計採用nlms演算法，通過設計自適應fir濾波器和語音檢測器達到回聲消除目的;對雙音多頻設計，信號發生端採用構造靜態參數表並通過二階正弦振蕩器產生信號，信號檢測端提取頻率信息以檢測信號;對呼叫進程音設計，除了類似雙音多頻的信號發生及頻率檢測設計外，還需要檢測信號持續時間，作者設計了一種基於匹配狀態表的方法以檢測信號持續時間。
  3. First, how to conduct sample and quantification of continuous time signal which is prior condition of sdr is explored in detail, and the comparison and analysis of some sample modes are given in which band pass signal sampling theorem is most important. second, multi - sample rate signal processing which is an important basis of sdr is studied. emphasis are put on decimation and interpolation those are the most fundamental process and the realization of decimation and interpolation filter

    在基於中頻采樣的軟體無線電結構框架下，首先詳細探討了軟體無線電的前提條件，即如何對連續時間信號進行采樣量化，比較分析了幾種采樣的方式，其中最為重要的是帶通信號采樣定理；然後探討了軟體無線電的一個重要基礎，即多采樣率信號處理，重點討論其最基本的兩個過程抽取和內插以及抽取器和內插器的實現；接著介紹了結構簡單、適用於一級抽取的cic濾波器和適用於做2倍抽取的半帶濾波器；再次論文在總結了傳統的調制解調基礎上，結合軟體無線電器件的特點，系統的探討並實現了基於正交思想的am 、 fm 、 ask 、 fsk 、 bpsk 、 qpsk的正交調制解調演算法。
  4. A novel communication receiver which uses lapped transform ( lt ) incorporating modified median filter ( mmf ) algorithm is designed for narrow - band interference excision. the lt domain mmf algorithm takes full advantages of the direct sequence spread spectrum signal, as well as the characteristics of lt, performing the transform domain filtering twice. the first filtering locates the position of interference and mitigates most of them. the second filtering is performed in a small neighborhood of the located interference. so lt domain mmf algorithm can completely mitigate the interference without distorting the desired signal. simulation results demonstrate the improved ber performance and increased robustness of our receiver

    本文採用改進的重疊變換域中值濾波演算法進行變換域抗干擾處理.該演算法既考慮到直接序列擴頻信號的特點，同時又利用重疊變換的特性對變換域系數進行了二次濾波.該演算法節省處理時間，並且不需要有關干擾的先驗知識，系統性能不會隨干擾頻率變化而變化，因而是一種很穩健的處理方法
  5. The paper first reviews the research background and actuality of the filter " s design in china and other country, introduces the meaning of the project and the work of the paper, narrates the theory of the switched - capacitor network and the basic switch building blocks, analyses the related factors of the design of sc filter. such as the selection of the architecture, the trade off of the opamp " s gain, bandwidth, phase margin, slew rate and setting time, the effect of the switch " s on resistor, how to reduce the charge injection and the clock feed - through, the power consumption and the selection of the sampling frequency and so on

    本文首先回顧了濾波器設計的國內外研究背景和現狀，介紹了本課題提出的意義以及本文的主要工作，論述了開關電容網路原理和基本開關模塊，分析了開關電容濾波器設計的相關因素：電路結構的選擇，對運算放大器設計中高增益、寬帶寬、相位裕度、轉換斜率和建立時間等的折中考慮，開關的打開電阻對電路的影響，開關電容電路中怎樣減少電荷注入和時鐘饋通，以及整個電路的功耗問題和采樣頻率的選擇等。
  6. In this thesis, first, we present the theory of sess system, the generation of the sess spreading code and its characteristics and the acquisition theory of conventional spread spectrum communication system. an efficient acquisition scheme based on periodically transmitting the synchronization head, which is composed of binary chaotic codes, using the matched filter and automatic decision threshold - level control based on a so - called constant false alarm criterion for sess system is present. the acquisition model of sess system is built and simulated in the awgn channel, the raleigh fading channel and imulti - address interfere condition

    本文首先概述了自編碼擴頻通信的原理、自編碼擴頻序列的產生方法及其特性和擴頻通信系統編碼同步的理論，然後針對自編碼擴頻通信系統提出了擴頻序列捕獲方案:周期性地加入混沌序列同步碼，並採用恆虛警率匹配濾波器捕獲法;在加性白高斯噪聲通道、瑞利衰落通道和多址干擾情況下進行了模擬，分析了各種捕獲性能:在選擇性能最優的混沌序列、適當的序列長度、虛警概率及門限值的情況下，可以獲得較短的捕獲時間和較大的捕獲概率。
  7. Though there are many methods in the field of speckle filtering, it lacks deep study of methods selection with different images and in different atr missions

    本文對採用空間域常用的濾波演算法、時間頻率域相結合的濾波演算法濾除sar圖像噪聲的結果進行了比較。
  8. Parallel structure of poly - phase decomposition and parallel mixer is applied in the ddc circuit, it solves the bottleneck in mixing and increases the handle speed. the partition of the tuning channel according to the digital mixing sequence, and the ddc by means of decimating first, the low - pass filtering and mixing realize efficiently the down - conversion of the variable carrier frequency band - pass signal. according to the structure of the ddc and the requirement of the frequency

    短數據快速測頻演算法的具體實現:使用并行流水線的設計方法，提高了系統的數據吞吐率，在100mhz的系統時鐘下，能夠實時處理400mhz ~ 600mhz速率a / d采樣的數據，在64點采樣， 100mhz系統時鐘情況下，初次測頻佔用時間640ns ，以後每次測頻佔用時間縮短到160ns ，實時地提供多相濾波下變頻所需的載頻位置信息，縮短了接收機的調諧時間。
  9. The following algorithms have been proposed and tested in the thesis : 1 frequency selective fading : combine the isomorphism between the input space and the output space and propose a new approach to blind equalization of the channel. compared with conventional methods, the new approach offers lower computational complexity, better performance, and more robust against the over - determination of the system order ; 2 time selective fading : a new approach to the equalization of time selective channel based on the zero - forced equalizer is proposed which is more simple in its structure of algorithm ; 3 time - varying channel : using the instantaneous mean value changes of the output signal to extract the information of channel variations and model it using ar model, kalman filter is then employed to track channel variations, it bears faster ability in tracking the variation of tv channels ; based on the isomorphism between the inputs and the outputs and some of the approaches using in mimo system, a new algorithm of equalization of simo time - varying channel is proposed, which also share the merits of being robust against the over - determination of the system order ; model the time - varying channel using the multi - resolution decomposition wavelets, and then a blind identification method based " on the model is proposed ; at last, a new model for equalization and identification of mimo system is proposed

    主要工作在以下幾個方面： 1 、針對頻率選擇性衰落通道：結合輸入輸出空間同構關系提出一種新的頻率選擇性通道均衡方法，與傳統方法相比，該方法計算量更小，收斂速度更快，性能更優，且對系統階次的過確定表現穩健，具有實際均衡應用價值； 2 、針對時間選擇性衰落通道：提出一種基於迫零均衡的時間選擇性通道均衡方法，演算法結構簡單； 3 、針對時變色散通道：利用瞬態均值曲線提取通道時變信息，對之ar建模，利用卡爾曼濾波器跟蹤時變通道抽頭變化，可以快速跟蹤通道變化；基於輸入輸出空間之間的同構關系以及多輸入多輸出系統的處理方法，提出了新的單輸入多輸出色散時變通道均衡與識別演算法，同樣具有對通道階次過確定保持穩健的優點；結合小波多解析度分析提出一種基於小波模型的通道盲識別演算法；研究時變的多輸入多輸出系統的盲均衡與盲反卷積問題，給出一種時變系統處理模型。
  10. In the fifth chapter, the performance of transconductor - capacitor ( gm - c ) continuous time filter is discussed. due to process variation and parasitics, an automatic tuning is designed for center frequency and quality factor q. also, in this chapter, a two order bandpass filter with tunable is designed. the effects on filter ' s performance of the non - idealities of a cmos ota are studied and the computer simulations at the mos transistor level are carried out

    第五章討論了跨導電容連續時間濾波器的性能特點，設計了一個中心頻率可調的二階帶通濾波器，為了使濾波器參數自動調整到設計標準值，從而保持其設計值的實現精度，論文給出了片內自校正（可調諧）環節。
  11. Computer simulation results show that the beamspace wsf algorithm retains the super performance of its element space counterpart when applied to the beam outputs of some practical acoustic - receiving array. 3. an improved form of classical time domain broadband beamforming is proposed by combining digital delay lines and fir filters

    提出了對經典時域寬帶波束形成器的改進方法，採用數字延遲線和fir數字濾波器結合的方式實現時域波束形成，以消除波束畸變並實時實現結構特殊的空間頻率響應。
  12. The design process of the multi - beamforming is presented detailedly, which include the orthogonal demodulation and resample technology, the schema of phase shift beamforming, and the effect of the frequency shift and the phase disturbance. the secondary filter is designed to eliminate the influence of the close frequency

    討論了頻偏和相位擾動對波束的影響；雙頻或三頻同時工作時，為消除相鄰頻率間的干擾，設計了二次濾波的特殊處理方法，模擬表明設計滿足要求。
  13. And the implementations in fpga of each composition are also completed. the rapidly frequency estimation algorism in the short data condition are implemented in fpga to get the band position. pipeline has been used in this dissertation to advance the performance

    結合多相濾波下變頻結構、演算法對測頻精度及速度的要求，提出了短數據快速測頻演算法的具體實現，使用流水線的設計方法，提高了系統的數據吞吐率，在盡可能短的時間內提供多相濾波下變頻所需的載頻位置信息。
  14. It can separate the real signal from noise signal effectively and attains the purpose of de - noise by using time - frequency localizing anslysis, local feature abstract, time - change filter wave, restraining or attenuating some frequency range and other characters of wavelet. furthermore, the ratio of signal to noise has been improved greatly and it is not affected by signal state

    而利用小波分析的時頻局部化分析、局部特徵提取和時變濾波、抑制或衰減某些頻率區間等特點，能更有效地把真實信號和噪聲信號區分開來，達到消除噪聲的目的，並且這種方法恢復的信號，信噪比有明顯的提高，還不受信號的狀態的影響。
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