頻率抽樣 的英文怎麼說

中文拼音 [bīnchōuyàng]
頻率抽樣 英文
frequency sampling
  • : 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...
  • : 率名詞(比值) rate; ratio; proportion
  • : 動詞1 (把夾在中間的東西拉出; 提取) take out (from in between) 2 (從全部里取出一部分; 騰出) ...
  • : Ⅰ名詞1. (形狀) appearance; shape 2. (樣品) sample; model; pattern Ⅱ量詞(表示事物的種類) kind; type
  • 頻率 : frequency; rate
  • 抽樣 : [統計] sample; sampling; specimen; samples draw
  1. In addition, a low - pass filter can be employed prior to the digital - to - analog ( d / a ) stage to smooth the stairstep effect resulting from the combination of a low sampling rate and quantization

    另外,在數模轉換( d / a )級前用低通濾波器可以平滑因低的和量化造成的階梯效應。
  2. To utilize the advantages of dsp chips, the system should be computing sources economical. according to digital signal processing theory, the poly - phase fir can help reduce the workloads of the ddc / duc. therefore, adding the complex carrier mixers, the channelization system ( a method of using a single wideband facility to transmit many relatively narrow - bandwidth signals. by subdividing the frequency spectrum used in the wideband channel ) can be formed utilizing the characters of fft

    為了使開發出的軟體可以適用於高速dsp器件開發,節省系統資源,課題首先從數字信號處理的理論進行分析,得出可以利用轉換的數字濾波器的特點,即多相濾波實現數字上下變計算負擔的減小,之後進一步將多相濾波器與譜搬移部分結合,通過公式的推導,得出可利用快速傅立葉變換的特點實現多路信號的通道化發射和接收的處理模型。
  3. Fpga and dvb standard are introduced firstly, dvb - c standard and composition of its system are analyzed completely, development of modulator structure and dvb - c digital modulator composition are presented, more over, analysis of respective modular are given. then, principle of dvb - c digital modulator system are presented, they are error control technique 、 mqam 、 nyquist rule and root raised cosine filter 、 window design method for fir filter 、 multi - rate signal processing ( integer interpolating, conversion of fractional sampling, equal conversion of net structure, polyphase structure for filter, poly - phase structure for interpolator, multi - stage implementation of samplying conversion ) 、 distrubited algorithm 、 cic filter 、 dds 、 cordic algorithm

    接著,專門利用一個章節闡述了dvb - c前端調制系統原理,他們了差錯控制技術、多進制調制( mqam ) 、 nyquist準則與平方根升餘弦濾波器、有限沖擊響應濾波器的窗函數設計法、多信號處理包括(整數倍內插原理、分數倍轉換、網路結構的等效結構、濾波器的多相表示、內插器的多相表示、轉換的多級實現) 、分散式演算法、 cic濾波器、直接數字合成( dds ) 、 cordic演算法。
  4. This thesis concentrates on the analysis of sampling rate changing ( decimation and interpolation ) ; the discussion of basic multirate signal processing theory ; the implementation of high efficient algorithm, which provides the bases for other algorithm development in this thesis

    本文從域的角度深入分析了變換的規律,並進一步研究了多系統的高效實現方案和基本理論,為其它演算法的研究提供了必要的基礎。
  5. In video shot segmentation, an improvement to double - threold shot segmentation algorithm is provided, which uses multi - frame sampling technique and can improve the performance significantly on the detection of gradual transition. an abrupt transition detection algorithm is also developed on the basis of the closest pixels matching in spatio - temporal slice, which decreases the false rate and computing strength greatly

    在視鏡頭分割方面,提出了一種基於多幀的雙重比較鏡頭分割演算法,有效地提高了對視鏡頭漸變檢測的性能;同時,針對視鏡頭突變的檢測,提出了一種基於最近鄰像素匹配的時空切片鏡頭突變檢測演算法,該演算法顯著降低了突變檢測的虛檢和計算量。
  6. This paper presents a method of realization of numeric transducer using tms320c542 dsp. the program of numeric down - transducer and multi - rate sampling is given

    摘要在tms320c542數字信號處理器上,用查表方法實現了dsp數字變器,並給出相關的數字下變和多信號處理程序。
  7. The bottleneck of hybrid filter banks adc system is that it cannot sample directly higher radio frequency signal because of lower analog input bandwidth of its adc. in order to remove it, a kind of downsampler model based on nyquist and bandpass sampling theorem is presented, analyzed and proved in time and frequency domain, in addition a downsampler is designed according to the model. on the basis of hybrid filter banks adc system, a class of high speed hybrid filter banks adc system is proposed

    針對混合濾波器組adc系統因其adc模擬輸入帶寬低而不能對較高的射模擬信號進行模/數轉換的瓶頸,作者提出了一種基於nyquist采定理和帶通采定理的取器數學模型,對該數學模型進行了時域、域的分析證明后,設計了一種基於該數學模型的sha取器,進而在混合濾波器組adc系統的基礎上,提出了高速混合濾波器組adc系統。
  8. 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。
  9. Digital audio aliasing is introduced when one attempts to record frequencies that exceed half the sampling rate

    在錄音超過的一半時,會造成數字音混疊失真。
  10. From theoretical analysis, we know the existing demodulation methods have limitations as following : one is that the subtraction of the two signals frequencies will display as the result of demodulation when we demodulate two time - domain adding signals without modulating information ( fault information ) ; the other one is that aliasing phenomenon will occur as a result of getting absolute value, detection or square in the process of generalized demodulation analysis, such phenomenon will result in some superfluous frequency composition on the frequency spectrum, which will puzzle the detec tion of mechanical vibration. if the sampling frequency is selected from a suitable range, the aliasing phenomenon will be avoided ; the last one is that aliasing frequencies will be produced in zoom demodulation analysis because this algorithm cannot employ digital low - pass filtering to avert the folding frequencies of higher harmonics in the process of zoom sub - sampling

    現有的解調分析方法存在以下三種局限性:將不包括調制信息(故障信息)的兩時域相加信號,也以其之差作為解調信號而解出;廣義檢波濾波解調分析中,由於取絕對值、檢波或平方過程可能產生混效應,在解調譜中表現為無法分析的成分,並由此推導出避免這種混現象的采的選取范圍,從根本上避免此類誤診斷的產生;幾種細化解調分析新演算法中,因為無法在細化分析的選時進行數字低通濾波,有可能會出現調制的高次諧波成分發生混疊而反折到低部分的現象。
  11. In a natural visual world dominated by low spatial frequencies, fixational eye movements appear to constitute an effective sampling strategy by which the visual system enhances the processing of spatial detail

    在由低空間所支配的自然視覺世界中,凝視時眼球運動似乎構成一種有效的視覺,以此提高了視覺系統對立體視覺的處理能力。
  12. The quality of digital audio is characterized by the sampling rate, the quantization interval and the number of channels

    數字音的質量可以用、量化電平和聲道數來表徵。
  13. It is found that the noun frequency correlates with preposition frequency, that the prepositions in nominalization are used in various ways, and that the prepositions tend to express metaphoric or abstract meanings

    研究發現,名詞與介詞相關,介詞在名詞化句型里用法多化,名詞化句型里的介詞更傾向于表達隱喻象意義。
  14. Except the simple and usual filter and algorithm, some advanced filter ways such as frequency sampling filter, karlman filter and several new algorithms such as cosine algorithm, semiwave fourier algorithm and wavelet transform algorithm are also studied

    除簡單、常見的濾波和演算法外,本章還探討了一些性能先進的濾波方法如頻率抽樣濾波、卡爾漫濾波和幾種微機保護新演算法如餘弦演算法、半波傅氏演算法以及小波變換演算法等。
  15. Objective to establish a complex pcr method and to investigate the genetic polymorphism and population difference of three loci on y - chromosome, so as to provide a database for forensic medicine casework. methods edta - blood specimens were collected from 163 unrelated males in han population in taiyuan. different tissues of one corpse were analyzed including blood, muscle, liver and kidney

    方法( 1 )本採集:隨機取163名太原地區漢族無親緣關系的男性個體的靜脈血, edta抗凝,進行dys390 、 dys391和dys393基因座等位基因及單倍型分佈調查;採集同一屍體血液、肌肉、肝臟、腎組織進行同一性檢測;採集20例兩代家系血進行突變觀察;取20例女性個體血進行男性特異性檢驗。
  16. The multiresolution method is extended and applied to random vibration control system, thus a multiresolution random vibration control algorithm is created, which provides a satisfied solution to the poor control precision problem at low frequency that was pointed out and remained unsettled for many years. the control algorithm has been successfully implemented on dactron ' s shaker control system, showing great improvement in low frequency control precision, with a little more computation resource consumed. high resolution octave analysis is another successful application of multiresolution spectrum estimation theory

    在多信號處理理論和多分辨譜估計理論的基礎上,本文進而將多解析度思想推廣到隨機振動控制理論中,提出了多分辨隨機振動控制演算法,解決了振動控制界多年來懸而未決的低控制精度問題,並在dactron公司laser ~ ( tm )和comet ~ ( tm )等多種振動控制產品中得到了成功的應用,在計算量增加不大的前提下取得了令人滿意的控制效果。
  17. The higher the sampling rate, the more bits per sample and the more channels means the higher the quality of the digital audio and the higher the storage and bandwidth requirements

    越高、每個的比特數越多、聲道越多,數字音的質量越高、所要求的存儲和帶寬越高。
  18. Integrating multirate signal processing theory and classical spectrum estimation theory, a brand new multiresolution spectrum theory is provided and a special algorithm is discussed in great detail. the polyphase decomposition theory is also applied to optimize the algorithm and improve the computing efficiency

    本文結合多信號處理理論和傳統譜估計理論,提出了嶄新的多分辨譜估計理論,並給出了具體的實現演算法,解決了譜估計中如何達到解析度和時間解析度的最佳組合的問題。
  19. All sampling requirements ( sample frequency, quantity, checking parameters, etc. ) well specified and documented

    所有要求(,數量,檢驗專案及參數等)都有詳細說明和有文件證明
  20. Finally, i simulate the performance of ldpc codes with different parameters on the uwb system. we find that the performance of ldpc codes is very good. we believe ldpc codes will become the main channel - coding scheme in the future uwb system

    最後,將ldpc碼應用於ds - uwb系統,並針對不同碼長、碼、用戶數、、訓練比特數進行了計算機模擬,證實了ldpc碼具有良好的糾錯性能,必將成為未來超寬帶uwb系統主要的通道編碼方案。
分享友人