original signal 中文意思是什麼
original signal
解釋
原始信號-
Experimental result shows that for sonant part of speech signal, 3 ~ 5 common ridges is enough to describe the main characteristics. signal compression is achieved by choosing proper way to represent the ridge information and use it to reconstruct the original signal
在信號重建過程中，選擇合適的方法用少量數據來描述起關鍵作用的參數，並用這些參數來重建信號，可以達到信號壓縮的目的。 -
With the auditory model as the front - end to extract the correlogram of signals. following, this paper present the implementation of suditory model inversion procedure by resynthesizing original signal from the correlogra - m
接著，文章闡述了通過實現聽覺模型反演過程從信號的自相關圖譜中恢復出原始的語音信號的過程。 -
Two block time - recursive algorithms are developed for the efficient and fast computation of the 1 - d rdgt coefficients and for the fast reconstruction of the original signal from the coefficients in both the critical sampling case and the oversampling case. the two algorithms are implemented respectively by a unified parallel lattice structure. and the computational complexity analysis and comparison show that the proposed algorithms provide a more efficient and faster method for the computation of the discrete gabor transforms
首先論證了一維rdgt系數求解演算法和由變換系數重建原信號演算法，不論是在臨界抽樣條件下還是在過抽樣條件下，都同樣具有塊時間遞歸特性，並提出了相應的塊時間遞歸演算法及其并行格型結構實現方法，計算機模擬驗證了并行格型結構實現的可行性，計算復雜性分析與比較也說明了rdgt塊時間遞歸演算法的并行格型結構在計算時間方面所具有的高速和高效性能。 -
In new system the error pa can also amplify the original signal, so the performance of whole amplifier system increases. since rf predistorters used in the new system, the linearity and power efficiency can be enhanced
該技術改變了誤差通路中的信號特性，使誤差放大器也輸出有用信號功率，提高了整個放大系統的性能。 -
It was indicated that the major frequency of signals for three different motivating manners concentrated between 20 and 50 khz, the difference between wavelet coefficients was small, and the pertinence of the original signal and the returning signals was low
最終結果表明，從頻譜上來講，三種激勵方式所產生信號的主頻集中在20 50khz ，小波系數相差甚微，並且來源於同一源的接收信號與源信號相關程度也不高。 -
Stationary phase method shows that by using information along the ridge defined by the phase on transformed domain, original signal could be represented and reconstructed
利用穩定位相法的近似公式分析可知，脊上分佈參數包含的信息與信號瞬時頻率變化規律密切相關，並且可以用來對信號進行重建。 -
The theoretical analysis and simulation show that the ssnr and the sksnr are equal to or greater than the original signal - to - noise ratio ( snr ), and that employing third order cumulant slices can effectively extract the mot - radiated quadratic phase dynamic coupling signals from asymmetrical distribution noise
理論分析與模擬表明：切片信噪比、歪度信噪比總是大於相關信噪比；用該演算法十分有效地實現了二次相位耦合信號與非對稱分佈色噪聲的分離。 -
The core of floating threshold value system is the data sampling technology. we will introduce the key chip - tlc5540 and tlc5615 and utilize mcs51 to control the whole system in the article. it examines the original signal from the hgcdte sensor and realizes the control of floating threshold value
浮動閾值系統的核心是數據採集技術，文中介紹了核心晶元tlc5540和tlc5615 ，利用mcs51單片機控制整個系統，對hgcdte傳感器輸出的原始信號進行檢測，實現了智能浮動閾值控制；同時闡述了性能先進的核心晶元? ? ad603及其應用於agc放大器的特殊性。 -
In many fields, such as remote sensing, nuclear medical, computer tomography, geophysical prospecting, groundwater hydrology, global ocean modeling, image processing, tumor detection, nondestructive detection, the objective of applied sciences and engineering is the recovery of the original signal given a collection of noisy observations of the original signal ; we call these problems inverse problem
在遙感技術、核醫療技術、計算機斷層掃描（ ct ）技術、地質勘探、地下水文學、電磁輻射、圖像恢復和重構、腫瘤檢測、非破壞性檢測等科學研究和工程實踐中，經常需要根據一組觀測到的數據來估計目標的真實信號，而觀測到的數據往往是真實信號經模糊、失真和加噪等過程后所得的輸出信號，通常稱這類問題為逆問題。 -
Once stored, the signals can be transcribed back into an exact replication of the original signal whenever desired. ultra high speed data acquisition is the main technique in drfm. the most common use of the drfm is for replication of complex, coherent signals usually representing radar return signals
數字射頻存儲器是現代電子對抗系統中有源雷達干擾機的主要組成部分，用於將接收到的雷達信號的精確的復制信號返回該雷達系統，以此來混淆該系統。 -
The statistic of wavelet transform coefficient algorithm can solve the periodic noise, high - energy noise and some non - gauss noise simply and effectively ; bi - spectrum can acquire more information from the original signal than power - spectrum, detect more information except from range and restrain the gauss noise. short - time speech signal can be considered as stationary and with periodic non - gauss signal, so we can make use of bi - spectrum to obtain the speech character and separate the speech and noise and detect morse telegraph signal ; complex number spectrum variance algorithm is put forward based on the deeply observing speech data, it is a new algorithm, experiment show that it is simple, effective
統計演算法在解決周期信號、高能噪聲和高斯信號方面有獨特之處，能簡單有效提取以上噪聲的特徵;雙譜能夠提供比功率譜更多的有用信息，有效地檢測信號幅度之外的其它信息，並能有效抑制高斯噪聲，短時語音信號一般認為是平穩且有一定的周期性的非高斯信號，因而可以利用雙譜來提取語音信號特性並實現信噪分離;復數譜方差演算法是在對語音信號進行深入觀察和分析的基礎上而提出來的一種全新的語音特徵提取方法，此方法簡單而有效的提取了語音、噪聲的特徵以及檢測莫爾斯信號，基於實驗表明，該演算法取得了很好的效果。 -
A " brick - wall " low - pass filter can remove the images and leave the original spectrum, thus recovering the original signal from the samples
這種頻譜的重疊導致的失真稱為混疊，而重建出來的信號稱為原信號的混疊替身，因為這兩個信號有同樣的樣本值。 -
This method reconstructs the noise signal by choicing some high frequencies of the original signal. of course, the original signal is decomposed by wavelet packet as low frequence and high frequencies. we used the matlab toolbox to demonstrate the method
該種方式利用小波（包）對含噪信號進行分解，然後選用某些高頻成分來重構滿足要求的噪聲，再對重構的噪聲信號進行采樣來生成隨機序列。 -
An effect that occurs when a signal is sampled at a rate less than twice the highest frequency present in the signal. the subsequent signal recovered from the samples will not contain the high frequency component of the original signal and will display a false low frequency signal
以低於信號中最高頻率兩倍的頻率進行信號取樣時出現的一種效應，即當由取樣狀態還原時，還原后的信號將不再含有原來信號中的高頻成分，並將顯示出虛假的低頻信號。 -
The usual way to solve this problem is to transform the original signal into " 0 ", " 1 " equally - distributing circuitry code type which is fit for the optical channel, and then transmit
解決該問題常用的方法是通過通道編碼將原始信號變換為「 0 」和「 1 」均勻分佈的適合光線通道傳輸的線路碼型，再送入光路進行傳輸。 -
In this paper, we propose an adaptive wavelet transform which possess the properties of translation and scale invariance. firstly, the original signal is adaptively renormalized using a scale function of an orthonomal wavelet and the first two moments of the signal. then, we decompose the renomalized signal according to the conventional discrete wavelet transform. as we prove, this adaptive wavelet transform is translation - and scale - invariant, and an efficient algorithm for calculating these wavelet coefficients, called adaptive wavelet invariant moments, is proposed. finally, we give experiment results for 2 - dimension digital signals ( images ) to verify our conclusion
本文提出了一種具有平移和尺度不變性的自適應小波分解新方法，該方法利用信號的一階、二階矩及正交小波尺度函數，先對信號進行自適應小波「重整」 .然後再對重整后的信號進行普通小波變換.本文證明這種自適應小波變換是平移和尺度不變的，並給出了計算自適應小波變換系數（稱為小波不變矩）的一種有效演算法.對二維數字信號（圖像）的實驗證實了我們的結論 -
Abstract : in order to process signal in depth and to extract the fault feature from original signal in machinery diagnosis ， the graphical display algorithm that can keep the data length of wavelet transform results the same as that of original signal is used. the fault diagnosis of a bend axial piston pump via b - spline wavelet that has linear phase is provided. the results of study demonstrate that the new method has excellent feature and the weak fault signal can be extracted from the strong vibration background of the pump
文摘:採用具有線性相位的b樣條小波，應用圖形顯示演算法並結合小波變換快速演算法將信號分解到不同的頻帶上，且分解結果和原信號長度保持一致，可對信號進行深層次的處理，克服了傳統的信號處理方法不易提取微弱信息的不足.通過對礦用斜軸式柱塞泵振動信號的分析，分離出了配流副磨損、球鉸松動、缸體及泵軸支撐軸承的故障特徵，為液壓泵的故障診斷提供了依據，並為信噪分離、微弱信號提取及設備的早期故障診斷提供了一條有效途徑 -
Per channel, and the result from ti ' s software simulator in ccs ( code composer studio ) is given, putting forward the principles and difficulties of realization of this algorithm. the recovered music signal is very close to the original signal and they are difficult to tell apart. in this paper a scheme of real - time implementation for this algorithm is discussed
本文敘述了mpeglayer音頻壓縮編解碼器的演算法及模擬實現研究，用c語言實現了的高保真音樂信號64kbps每聲道的非實時解碼，並在ti的ccs （ codecomposerstudio ）系統中的軟體模擬器上進行實時研究，提出了該演算法在具體實現中的要點和難點。 -
We are usually it is really wave that what say lose lose true, point at the voice return to put of process inside, increased the original signal the composition of high wave that have no but cause of lose really
我們通常所說的失真是諧波失真，指在聲音回放的過程中，增加了原信號沒有的高次諧波成分而導致的失真。 -
The information of singularity points can be reserved well and the de - noised signal is a good estimation of the original signal
該方法在去噪的同時，能有效地保留信號的奇異點的信息，是原始信號的一個非常好的估計。
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