過量噪聲法 的英文怎麼說

中文拼音 [guòliángzàoshēngfǎ]
過量噪聲法 英文
excess noise technique
  • 過 : 過Ⅰ動詞[口語] （超越） go beyond the limit; undue; excessiveⅡ名詞（姓氏） a surname
  • 量 : 量動1. （度量） measure 2. （估量） estimate; size up
  • 噪 : 動詞1. （蟲或鳥叫） chirp 2. （大聲叫嚷） make noise; make an uproar; clamour
  • 法 : Ⅰ名詞1 （由國家制定或認可的行為規則的總稱） law 2 （方法; 方式） way; method; mode; means 3 （標...
  • 過量 : excessive; overdose; surfeit; excess; overmeasure; overage; bellyful
  1. The dissertation discusses the characteristics of the human auditory system and its describable method and with this understanding establishes the objective method based on subjective perception. it is clarified that the person ' s evaluation on noise is not only " subjective " and qualitative analysis, but also quantitative analysis based on stable foundations. the availability and some problems of these methods of sound quality are illustrated with some concrete emulational examples

    論文詳細討論了人耳聽覺的一般特性及其描述方法，以其為基礎確立聲質量的基於主觀感覺的客觀評價方法，說明了人對噪聲的評價並不僅僅只是「主觀」的、定性的東西，而是可以建立在堅實基礎上來進行定量分析的，並通過具體算例說明聲質量方法的有效性和存在的問題。
  2. During simulation of kinematic calibration of virtual axis machine tool, indirect method is used for measuring its position and orientation ; meanwhile jacobean matrix regrouping method is used instead of arduous optimization to avoid measurement noise maginification

    摘要在對虛擬軸機床進行運動學標定的模擬計算過程中，採用間接測量法對虛擬軸機床的運動進行間接測量，然後通過矩陣重構的方法來解決測量噪聲的干擾問題。
  3. Backed up by the soplat theory based on particle kinematics, the second chapter of this paper presents with analysis and simulation of several single observer passive measurement models, which uses such relative movement parameters as bearings changing rates and centrifugal acceleration information on the basis of bearings measurements. in the third chapter, the observability of location respectively using bearings and its changing rates information and centrifugal acceleration information is analyzed, and its observable condition is got. the fourth chapter puts forward the modified covariance extended kalman filtering ( mvekf ) against the defect of traditional extended kalman filtering ( ekf ), whose performance is simultaneously compared in the chapter with the performance of ususal tracking algorithm such as ekf, mgekf, iekf by computer simulation

    在近年來提出的基於質點運動學原理的單站無源定位理論基礎上，本文第二章提出了幾種在角度測量的基礎上增加角度變化率及相對運動的離心加速度等運動學參數的單站無源測量模型，並對它們進行了分析和模擬；第三章分別對利用角度及其變化率信息定位和利用離心加速度信息定位的可觀測性進行分析並得到了相應的可觀測條件；第四章針對傳統擴展卡爾曼（ ekf ）方法的缺點，提出了一種修正協方差的擴展卡爾曼濾波（ mvefk ）方法，並將其和ekf 、 mgekf 、 iekf等常用的單站無源定位濾波方法進行了性能模擬比較；第五章通過引入雷達機動目標跟蹤方法和模型，提出了利用角度及其變化率對機動輻射源跟蹤的多級噪聲自適應方法和imm方法；第六章主要對角度變化率和離心加速度參數的獲取技術進行了研究，提出了幾種高精度測量脈沖序列多普勒頻率變化率的方法。
  4. This paper studies the mechanism generated from flow - noise in seawater pipeline, introduces controlling method of flow - noise, analyzes sound field characteristics of flow - noise in seawater pipeline system and the merits and demerits of measuring the flow - noise methods in and out of the pipeline. in our experiment, a test measuring device of seawater muffler is used in a measuring chamberlet to verify that a method of reverberation chamberlet is difficult to exactly measure the low frequency flow - noise in pipeline. at the same time the insertion lose of water muffler in the low frequency was measured by the measuring chamberlet method

    本文分析了海水管路系統內流噪聲的產生機理，介紹了流噪聲的控制方法；分析了海水管路系統中流噪聲的聲場特性以及流噪聲管內、管外測量方法的優缺點；通過海水消聲器試驗測量臺架，利用測量小室驗證了混響箱法難以準確測量海水管路系統中的低頻流噪聲；用測量小室對水消聲器在低頻段的插入損失進行了測量。
  5. Compared with the classical bp algorithm, robust adaptive bp algorithm possesses some advantages as following : ( 1 ) increasing the accuracy of the network training by means of using both the relative and absolute residual to adjust the weight values ; ( 2 ) improve the robustness and the network convergence rate through combining with the robust statistic technique by way of judging the values of the samples " relative residual to establish the energy function so that can suppress the effect on network training because of the samples with high noise disturbances ; ( 3 ) prevent entrapping into the local minima area and obtain the global optimal result owing to setting the learning rate to be the function of the errors and the error gradients when network is trained. the learning rate of the weights update change with the error values of the network adaptively so that can easily get rid of the disadvantage of the classical bp algorithm that is liable to entrap into the local minima areas

    與基本bp演算法相比，本文提出的魯棒自適應bp演算法具有以下優點： （ 1 ）與魯棒統計技術相結合，通過訓練樣本相對偏差的大小，確定不同訓練樣本對能量函數的貢獻，來抑制含高噪聲干擾樣本對網路訓練的不良影響，從而增強訓練的魯棒性，提高網路訓練的收斂速度； （ 2 ）採用相對偏差和絕對偏差兩種偏差形式對權值進行調整，提高了網路的訓練精度； （ 3 ）在採用梯度下降演算法對權值進行調整的基礎上，通過將學習速率設為訓練誤差及誤差梯度的特殊函數，使學習速率依賴于網路訓練時誤差瞬時的變化而自適應的改變，從而可以克服基本bp演算法容易陷入局部極小區域的弊端，使訓練過程能夠很快的「跳出」局部極小區域而達到全局最優。
  6. The first two methods use the advanced gaas materials and devices to reduce the noise of receiving system. without increasement of the transmitted power, the system margin gains 1 db compared to the conventional project margin, the information rate can be increased 2 mbps or more

    前兩種方法通過採用先進的gaas材料和器件降低接收系統的噪聲，在發射功率不變的情況下，使系統余量比常規工程設計余量提高1db以上，信息速率可以提高2mbps以上。
  7. The experimental results show that the gained curve using our algorithm is smooth and can capture original curve ' s character, remove the noise, represent drawer ' s intention. moreover, the points are relatively fewer that confirm the principal curve. all these suggested the practicality of the adaptive method in this paper

    通過大量的試驗證明，本文提供的演算法得到的主曲線能抓住原徒手畫曲線的特徵，光滑流暢，去除了原曲線的噪聲，反應繪圖的願望，並且確定該主曲線的數據點較少，充分體現了本文徒手畫曲線自適應方法的實用性。
  8. This thesis explains the necessity of the character recognition technology of the computer at first, describe the meaning in which the handwritten numeral discerns ; pretreatment technology of handwritten numeral recognition, including two value, line segmentation, word segmentation smooth, removing noising, standardization and thinning are discussed two value concretely discusses whole threshold value, some threshold value, dynamic threshold value and utilize space information to carry on threshold, which are several kinds of common method of choosing threshold value, especially utilize space information to carry on threshold value is describe in detail ; adopting to the foundation of thinning based on mathematics morphology, thinning algorithm of serials same and thinning algorithm of protecting shape are discussed ; afterwards, according to principle ' s diagram of the on - line character recognition, by analyzing the structure feature of the handwritten numeral, this thesis has proposed the online recognition te chnology of the free handwritten numeral based on the stroke feature and the online recognition technology of the free handwritten numeral based on the multistage classifying device. detail narrated noise removing, stroke characteristic definition and discernment, distance criterion of whole word match ; then under the foundation of handwritten numeral segmentation, off - line handwritten numeral recognition is researched. especially minimum distance classifying device, tree classifying device and adaptive resonance ( art ) network classifying device is discussed at the same time, believes degree analyses are introduced to integrate a lot of classifying devices ; at the end, the typical application of the handwritten numeral recognition was briefly narrated, its application in extensive data statistics, financial affairs, tax, finance and mail sorting have been explored

    二值化時對整體閾值二值化、局部閾值二值化、動態閾值二值化和利用空間信息進行閾值選取幾種常用的閾值選取方法進行討論，特別對利用空間信息進行閾值選取進行了詳細論述；在對通過對基於數學形態學的細化的基礎上，討論序貫同倫形態細化演算法和保形的快速形態細化演算法；然後依據聯機字元識別原理框圖，分析了手寫數字的結構特點，提出了基於筆劃特徵的任意手寫數字在線識別技術和基於多級分類器任意手寫數字在線識別技術，對其中涉及的筆劃識別前的噪聲處理、筆劃間特徵量的定義及識別、整字匹配的距離準則進行了詳細敘述；繼而在對手寫數字的分割的基礎下對脫機手寫數字識別進行了研究，對基於最小距離分類器字元識別、基於樹分類器的字元識別、基於自適應共振（ art ）網路的字元識別分別進行了詳細討論，並引入置信度分析將多個分類器進行了混合集成；最後簡單闡述了手寫數字識別的典型應用，對其在大規模數據統計、財務、稅務、金融及郵件分揀中的應用進行了探索。
  9. 4. the dissertation studies in - depth the multi - aperture sar mtdi based on along - track interferometry ( ati ). it presents and describes the detecting processing in sar image domain using double apertures and analyzes the influence of all kinds of factors including ground clutter, noise, signal - to - clutter, range velocity, multi - look processing and etc on the detection performance

    4 、深入研究了基於沿跡干涉釁1 ）技術的多孔徑sar系統地面動目標檢測和成像方法，給出了採用ati方法對兩幅sar復圖像進行慢速目標檢測的完整過程，詳細分析了各種因素如雜波、噪聲、目標信雜比、目標速度及多視處理等對檢測性能的影響，並通過大量的模擬給予驗證。
  10. By the line of cross - cut, the 2 - d fmm ii with multiplicative noises is transformed into the form of 1 - d model, then the proposed algorithm is obtained on the basis of state estimation theory of the 1 - d systems

    利用斜割支線法，通過定義新的變量，巧妙地將帶乘性噪聲的2 - dfmm模型轉化為一維帶乘性噪聲的狀態空間模型。
  11. Chapter 4 : the influence of different phase - shifting angle of reference wave in recording, of the phase - shifting error and of the quantization error on the quality of the reconstructed image is investigated respectively in two - step phase - shifting inline digital holography. and an effective method of eliminating the phase - shifting error is presented, in which the summation of the intensity bit errors of the reconstructed image is taken as an evaluation function for an iterative algorithm to find the exact phase - shifting value. the feasibility of this method is demonstrated by computer simulation

    通過數值分析不同相移角的選取對再現像的強度誤差的影響，發現再現像強度誤差依賴于記錄時相移角的選取，當相移角在一定范圍內，其再現像的強度誤差相對較小；提出了一種有效消除相移誤差的新方法，並將其應用於二步同軸相移數字全息，對此進行了計算機模擬，得到了很好的結果，證明該方法對于相移誤差的消除是很有效的；對量化誤差所作的數值模擬發現，當信號被量化成8比特（ bit ） ，即256個灰度級以上，量化噪聲的影響相對較小。
  12. Considered the tiny quantity of the hydrogen, first we choose the gas chromatography technology to analyze the component of the work gas and to obtain the chromatography curve. then, treat the curve with the adaptive aperiodic stochastic resonance algorithm in order to eliminate the apparatus noise submerged in the tiny hydrogen signal. at last, calculate the quantity of the hydrogen based on that the quantity varies directly as a function of the area of the chromatography curve

    考慮到氫的含量非常微弱，在測量時採用氣相色譜法，並通過自適應非線性隨機共振演算法對色譜信號進行處理，以便提取出被儀器噪聲淹沒了的微弱氫信號的色譜曲線，最後根據氫含量與其色譜峰面積呈正比計算出工作氣體中的氫含量。
  13. Under the condition of " comparatively weak correlation between the two noises involved, coherence function is used as a frequency domain amplification factor for improving snr of the output signal to the filter and the speech enhancement effect. meanwhile, a real - time recursive algorithm is put forward in substitute for current algorithms based on short time fourier transform. the new algorithm will simplify computations and will be suited for real - time implementation together with the adaptive systems

    接著針對上述nanc系統兩路輸入信號噪聲相關性弱的情況，用相干函數作頻域增益因子來提高輸出信噪比與改善語音增強效果，同時，通過一種實時迭代演算法解決了短時傅氏變換計算量大的問題，簡化了計算，便於實時處理與實際應用。
  14. On one hand, from the technique of constrained least squares and limited energy of additive noise, an effective restored approach by adopting regularization method to overcoming ill - posed problem, solving an equation with a single variable, and using space iterative algorithm is proposed ; on the other hand, aiming at the restoration of blurred image, another effective restoration approach based on least - square algorithm is also proposed in this paper. this method firstly adopts increment iterative algorithm to improve convergence and meanwhile applies regularization technique to overcome ill - posed problem. in the computations, the regularized parameter has its adaptive character, which can be determined in terms of the restored image at each iteration step therefore automatically correct to the appropriate value

    一是從約束最小二乘出發，在加性噪聲能量有界的前提下，採用正則化方法來克服病態問題，通過解一個單變量方程，並利用空域迭代運算實現了一種有效的圖像復原;二是針對模糊圖像的復原問題，從最小二乘演算法出發，採用增量迭代的方法改善演算法的收斂性，同時結合正則化技術克服問題的病態性質，並引入自適應的正則化參數，使其與圖像復原的迭代運算同步進行並自動修正到最優值。
  15. Fortunately, we can solve this problem by using ap ( alternating projection ) algorithm, by this way ; the computational load can reduce dramatically. doa estimation is mostly research base on linear array, because of the limitation of linear, it can estimate the azimuth only ; solid array can estimate the azimuth and elevation, in this paper, the solid array is used for doa estimation ; when the signal sources are uncorrelated, the music algorithm is chosen, using mdl / aic algorithm, we attain the signal number of sources

    本文採用立體六元陣進行測向，在非相干信號源條件下，採用music演算法進行doa估計，採用基於信息論原理mdl準則或者aic準則判斷信號源數目，通過對陣列流形的研究，指出基於圓陣的陣列流形具有的某種對稱性，利用該對稱性來降低運算量，並且通過對演算法的研究，提出用維數較小的信號子空間或噪聲子空間來計算空間譜，進一步減小music演算法運算量。
  16. Secondly, ground on the duct noise control, the delayed - lms algorithm, which is actulized easily, is brought forward, then the simulation on pc is achieved. it is shown from the computer simulation that for the single frequency noise, there will be a good effect when choosing the suitable parameters ; but for the narrowband noise, the effect will become worse with increasement of the bandwidth. this result is helpful to analyze the parameters influencing anc

    結果表明： （ 1 ）隨著濾波器長度的增大，系統收斂速度變快； （ 2 ）隨著值的增大，系統的收斂速度增大，但是當值超過某一范圍時會引起系統的發散； （ 3 ）次級通路的延遲數與初級噪聲頻率有關； （ 4 ）對于窄帶信號， dlms演算法可以取得一定的降噪量，但是隨著窄帶信號帶寬的增大，降噪效果明顯變差。
  17. The first part of the paper is designing the testing project for grounding resistance and insulation resistance in a new way. using 16bits ad converter with programmable control amplifier replaced the way which used changing resistance to change measure range. lt is not only improved testing precision and develop the system expediently, but also reduced the area of the circuit boardwith the new way. in order to make the electric implement safety testing system have upstanding expansibility, the software and hardware of the system adopted the modularization design. adopted mcu atmegal28 as a master mcu which control mmi, realtime clock and communication with slaver mcu. atemga8 as the slaver mcu to realize testing function. so it is easy to add or reduce the testing project. the testing implement system has been developed successfully, and the comments for the system is that it has high precision, high expansibility and easy maintain. but considering the electric implement system should have intelligence and humanity abi lity. so this paper bring forward a scheme of electric equipment safety testing embedded system with speech control. after introduce the basic theory of speech recognition, the paper expatiate the characters of this system. the system is a noise conditon, not special people, small glossary, insulation word system. with these characters design the speech recognition as fellow. utilizing cross zero ratio and short energy to ensure jumping - off point and end point ; adopting mfcc as the character parameters of speech recognition ; the character parameters than be recognized by dtw. in order to ensure the credibility of this project, first realized by matlab in computer

    在介紹了語音識別的基本原理后，闡述了本系統的特點：本系統是一個噪聲環境下非特定人、小詞匯量、孤立詞的語音識別系統。根據本系統的這些特點設計了如下語音識別方案：利用過零率和短時能量相結合的方式確定語音端點；採用mel頻率倒譜系數（ mfcc ）作為語音識別的特徵參數；得到的特徵參數最後通過動態時間規整（ dtw ）的模式識別方法進行識別。為了確保本系統實現方案的可靠性，首先通過計算機利用matlab軟體來模擬，在演算法模擬實現后又進一步增加環境的復雜性：加上較大的環境噪聲、突發性的噪聲等，再通過修改參數、修改參考模板、兩級識別等各種提高語音識別精度的方法來提廣東工業大學工學碩士學位論文高識別率。
  18. The traditional detection algorithm, based on zero - crossing or energy, will not acquire ideal effect when the signal - to - noise is low or the signal is weaker. therefore, to resolve the real problem in the real environment that all kinds of random noise and speech signal exit together, some new algorithm must be put forward. account for the complexity of real noise, we integrate the wavelet transform and high - order statistics and advance a new algorithm ; the algorithm can effectively separate the speech signal and the non - gauss noise

    基於過零率和能量的傳統檢測演算法，在噪聲環境比較復雜的情況下效果很不穩定，尤其是信噪比較低或者語音信號較弱時，檢測效果很不理想，因此，在多種語言和噪聲隨機出現、噪聲和語音強弱不一的實際噪聲環境下，必須利用新的演算法提取有用信號和噪聲信號的有效特徵，才能解決實際的問題。
  19. We present an image - hiding cryptosystem based on vector quantization and wavelet transform. the key information of secret image is encrypted and then hidden into another image, which can be publicly accessed. in the paper, we compress the key image information using vector quantization, and apply wavelet transform method in compressing and noise - removing to improve the image quality

    論文中，使用矢量量化的方法大大壓縮了保密信息的數據量，並將圖像的小波變換應用在圖像的前期壓縮和後期去噪聲的過程中，提高了壓縮比，增強了明圖的偽裝性，減少了運算時間，優化了解密圖像的效果。
  20. The apsp produced abroad is made by numerical controlled machine tool, which has noise level of 71db ( a ), the apsp produced in our country is made in the method of exploratory which has noise level of 73db ( a ) and 75db ( a ). in order to analyze the influence of stator curve to noise, the author used tri - coordinate measuring instrument to measured exactly the inside surface of stator and got the straddling point coordinate, and made curve fitting by using matlab as language and studied the fitting effect and then worked out the equation of stator transition curve in return seeking, then comparied this method with the standard style and made the conclusion : the equation of atator transiting curve of apsp made by numerical controlled machine tool is close to theorical 5 power curve standard style, but compared with it, the stator transiting curve of apsp produced in exploratory has a major error. combining the testing results of noise, one can know that the qualily of stator transition curve play an importance to t he noise of the pump

    國外生產的汽車動力轉向泵是用數控機床加工的，其噪聲值為71db （ a ） ，國內生產的汽車動力轉向泵是用靠模方法加工的，其噪聲值分別為73db （ a ）和75db （ a ） ，為了分析定子曲線對泵的噪聲的影響，本人用三坐標測量儀對定子內表面進行精密測量，獲得定子內表面的離散點坐標，以matlab語言為工具對離散點進行曲線擬合，觀察擬合效果，然後，用回歸方法求出了定子過渡曲線的方程，並把該方程與理論方程標準型進行了比較，得出如下結論：用數控機床生產的汽車動力轉向泵的定子過渡曲線方程非常接近理論5次曲線標準型，而用靠模方法加工的汽車動力轉向泵的定子過渡曲線與理論5次曲線標準型相比較，則存在著較大的誤差，結合噪聲測試結果可知，定子過渡曲線的優劣，對泵的噪聲大小有著重要的影響。
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