discrete-time signal 中文意思是什麼
discrete-time signal
解釋
離散時間信號-
Taking a view of pure time field, this paper presents dissolvable signals and basic concepts of sampling ; and also gives the very condition of discrete of signal ; this paper uses linear algebra to analysis this kind of signal and then gives some results and relevant deductions ; based on these results, 1 made a further step to analysis some typical band - limited signals in order to proof the coherence of my theory of dissolvable signal to the typical sampling theory ; i made some preliminary study about the feasibility of sampling and recovering of this kind of signal
本文從純時域角度出發，給出了可分解信號及其采樣的基本概念；也指出了信號可離散化的條件；利用線性代數理論給出了對這類信號進行采樣的分析理論及相應的推論；並用這些結論對典型的帶限信號進行了分析，證明可分解信號采樣定理與經典采樣理論的一致性；初步探討了對這類信號的實行采樣與恢復的工程實現問題。 -
This thesis tries to update the cmdsr system to achieve the characters below : real - time, better robust, higher recognition rate, non - special - man. considering the disadvantages of traditional improved spectrum subtraction speech enhancement, this thesis proposes the theory of fuzzy spectrum subtraction based on the fuzzy theory and improved spectrum subtraction speech enhancement ; as for the difficulties of detecting the endpoint of speech signal, the thesis gives the table of initial and the improved parameters, with which we can confirm the endpoints of mandarin digit speech ; the thesis puts forward two - level digit real - time speech recognition system, the first level is based on discrete hidden markov model which is linear predictive coding cepstrum ( lpcc ) and difference linear predictive coding cepstrum ( dlpcc ), the second level is based on formant parameters ; as for the realization of hardware, the thesis depicts the realization of every part of cmdsr based on the tms320vc5402 in detail ; as for the development of software, the thesis gives the software design flow chart of cmdsr, simulates the basic theory with matlab language and gives the simulation results
針對傳統的「改進譜相減法語音增強」參數設定單一、環境適應能力差的缺點，提出了一種利用模糊理論和「改進的譜相減法」結合的「模糊譜相減法語音增強」 ;針對語音信號端點檢測困難的特點，通過matlab模擬試驗，給出了能夠準確確定數碼語音端點的初始和改進參數表;提出了利用基於線性預測編碼倒譜參數和差分線性預測編碼倒譜參數相結合的離散隱含馬爾可夫模型進行第一級識別、利用共振峰參數進行第二級識別的兩級漢語數碼語音識別系統，在保證系統實時性的同時，實現連接漢語數碼語音識別系統識別率的提高;在硬體實現上，詳細闡述了基於tms320vc5402的連接漢語數碼語音識別系統各部分硬體設計;在軟體開發上，給出了連接漢語數碼語音識別的軟體設計各部分的流程圖，並對各部分進行了matlab模擬，並給出了模擬結果。 -
This thesis has studied the dynamic features of a class of the discrete - time neural network model of two neurons, such as the convergence and periodicity and etc. the function of the neuron signal transmission in this model, which belongs to three piecewise constant argument, indicates the following charactersif the signal of one neuron on the network is active between a and b, it will produce invariable encouragement effect on another neuron ; if the signal of one neuron is lower than a, it will produce invariable restrain effect on another one, if the signal of one neuron is higher than b, it will produce no effect on another one
本文研究了一類二元離散人工神經網路模型的解的收斂性及周期解的存在性等動力學特徵。該模型的神經元信號傳遞函數是三段常數不連續函數。這種信號傳遞函數表明如果某神經元的信號在a與b之間活躍，則它對另一個神經元產生恆定的激勵效果，如果某神經元的信號低於a ，則它對另一個神經元產生恆定的抑制效果，如果某神經元的信號高於b ，則它對另一個神經元不產生作用。 -
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。 -
Fir ( finite impulse response ) filter is one of the basic algorithms for digital signal processing, which is a kind of important lti discrete - time system, widely used in acoustic processing and image processing area
Fir （ finiteimpulseresponse ）濾波器是數字信號處理的基本演算法之一，是一類較為重要的線性時不變系統，廣泛應用於聲音、圖像處理等現代通信技術中。 -
In this paper, researching and producing on current waveform control system of inverter carbon dioxide welding machine is given. the overall plan, the design of hardware structure, the programming and debugging of software is described in detail. the bang - bang and alterable structure discrete pid synthetic control algorithm are used to modulate the width of pwm signal in real time, to acquire the fiat character by driving and controlling the opening and shutting time of igbt, the technology of " double impulse " is adopted to realize electric current waveform control
本文介紹了逆變波形控制co _ 2焊機控制系統的研製，詳細闡述了總體方案的制定、硬體研製及軟體的編寫與調試，採用變結構、離散型pid + bang - bang復合控制演算法、實時調節輸出pwm信號的脈沖寬度，驅動控制用於逆變的大功率開關管igbt的開通與關斷時間，獲得逆變焊機平外特性。 -
We propose the joint filterbank precoders and decision feedback equalizers structure at first, by which the dispersive channel is equivallent into parallel independent flat fading subchannels such that the diversity gain of the receiver is increased. then we adopt the discrete - time ( dt ) canonical model to convert the problem of blind signal processing of tv dispersive channels into processing the time - invariant multi - channels model blindly, and discuss the problem of blind equalization and identification of tv dispersive channels based on this new model
第一種是採用濾波器組聯合均衡方法將色散通道等價為一組獨立的平坦衰落子通道，以提高接收機的分集增益；另外一種是提出採用離散正則模型將時變色散通道的盲信號處理轉化為時不變多通道模型的盲信號處理，並針對該模型對時變色散通道的盲均衡與盲辨識方法進行了詳細討論。 -
Secondly, a dstft ( discrete short time fourier transform ) - based demodulation method for the 2fsk signal is studied and algorithms for symbol synchronization and symbol decision are analysed
重點研究了基於離散短時傅里葉變換解調2fsk信號的方法，分析了幾種碼元同步演算法和碼元判決演算法。 -
One is to use fourier transformation to convert the source signal from time domain to frequency domain and to discard high frequency harmonious components upwards of 19 ( gb / t14953 - 93 d5. 3 demanding ), then to have static huffman coding to the quantized char array which is composed of reserved direct current component and basic wave and each high frequency " s amplitudes and angles. the other is to use discrete wavelet transformation to convert the source signal from time domain to frequency domain and to set the high frequency coefficients that its absolute value is smaller than the given threshold to zero, then to have dynamic huffman coding to the quantized char array which is composed of multiple, wavelet ' s level, datum length, low frequency coefficients and reserved high frequency coefficients. mass simulinks and analyses under the two circumstances have done to show that data compression ratio is small and the relative error is also small and within the permission of engineering and the compression problem can be solved in theory of measured datum of power system
第一種情況的壓縮方法為：採用傳統的傅立葉變換把原始信號從時間域變換到頻率域，舍棄20次及其以上的高次諧波成分（保證了gb / t14953 ? 93d5 . 3要求） ，然後對保留的直流分量、基波和各次諧波的幅值和相角數據量化后和量化時分別乘以的倍數系數構成一個數組，以字元形式保存，採用靜態huffman編碼對變換數據進行壓縮；採用離散小波變換把原始信號從時間域變換到頻率域，然後對分解得到的高頻系數進行閾值量化處理，對乘以的倍數系數、小波變換的階數、小波變換后的低頻、各級高頻以及原始數據長度、量化后的低頻系數以及保留的高頻系數大小、位置構成一個數組，以字元形式保存，採用動態huffman編碼對這個文件進行壓縮。 -
Based on the analysis of the theory, the paper put forward the method of the analysis and design of the sc filter by the discrete time signal method using the basic switch blocks combined with the fabrication convenience, mature technology, low power consumption, higher integrated cmos process
在理論分析的基礎上，論文提出了結合現在尋求代工方便、技術比較成熟、功耗低、集成度高的cmos工藝技術，利用常用的基本開關模塊，用離散時間信號的方法來分析和實現開關電容濾波器。 -
Secondly, a network based on multi - terminal components modeling methodology was applied to model mems at system - level by the analogy and mixed - signal modeling tool of vhdl - ams, for the system - level model of mems is a mixed signal model, which has attributes of multi - energy domains coupling, multi - signals mixed and interacting between discrete - event subsystems and continuous - time subsystems. with this method, the whole system can be divided into some subsystems defined as multi - terminal components ; the behavior of the subsystems depends only on their terminal signals ; the information exchange between subsystems was done by the signals at their terminals. the continuous - time systems or discrete - event systems can be modeled and simulated with this method, which satisfied the requirements of nonlinear systems and large signals analysis
同時，針對mems的系統級模型是一個混合信號模型，具有多能量域耦合、多信號混合、離散事件子系統與連續時間子系統交互的特點，使用vhdl - ams作為混合信號模型建模的工具，採用多埠組件網路建模方法建立了mems系統級模型，把微型機電系統分解為多個子系統或組件，各子系統被定義為多埠組件，子系統的內部行為通過其埠行為來描述，子系統間的能量與信號的交換通過組件的埠映射來實現，從而實現了對連續時間系統和離散事件系統的建模與模擬，滿足了非線性系統以及大信號分析要求。 -
The theory of soplat using frequency rate is discussed in chapter 2. based on this theory, this chapter studies the observed discrete - time signal. the crlb of continuous wave and coherent pulses frequency estimation are also studied
論文第二章討論了利用頻率變化率的單站無源定位原理，在此基礎上，介紹了單站無源系統中接收信號的模型，並分析基於各模型的頻率估計的cramer - rao下界。 -
Nowadays, there are two kinds of implementation methods of chaotic signal generator. one kind is realized by analog line and it is very sensitive to circuit inherent parameters as well as signal recycled error, so it is relatively difficult to realize actually, the other kind is realized by digital line and it can generate well real - time discrete array, so it is more suitable for the application in communication
目前混沌信號發生器的實現方法有兩種:一種是由模擬電路實現，它對固有參數及信號再生的誤差很敏感，實際實現較困難;另一種是由數字電路實現，它產生的離散時間混沌序列實時性好，更適合在通信中應用。 -
In voice print recognition, the production of the sound signal is usually modeled by the discrete time - domain model. with the respect to feature extraction, the requirement is that feature parameters selected should reflect both the physiological properties of the speaker ' s vocal organs and the psychological properties of the speaker ' s speech habits
聲紋識別中也是採用離散時域模型作為語音信號產生的數字模型，在特徵提取方面，特徵參數的選取要反映出說話人發音系統的生理特點和明顯反映出說話人發音習慣等的心理特點。 -
A modified real signal model of channelized transmitter is presented that employs multiphase filters and discrete fourier transform to maximize computing efficiency, which can mean much in a radio communication system that requires real - time processing
特別提出了改進的實信號通道化發射機數學模型，通過利用多相濾波與離散傅立葉變換，減少信號處理運算量，這對于強調實時運行的無線通信系統有重要意義。 -
By studying the discrete fourier transform properties of the band - limited digital signal, the authors introduce alternating projection neural networks into the paper, expand apnn ' s application scope from real field to complex field, and present several important conclusions on apnn. analyzing and discussing network ' s tolerance to noise, convergence rate and the spectral leakage problem of the truncated signal expected to be extrapolated by using these conclusions, the paper presents an extrapolation algorithm for band - limited signals based on alternating projection neural networks. a lot of simulation experiments show that the algorithm is effective. in addition, the algorithm is also effective to spectrum extrapolation. owing to adopting network structure, the algorithm is prone to parallel computation and vlsi design, and consequently can satisfy real time military processing needs
本文通過對頻帶受限數字信號的離散傅立葉變換特性的研究，引進了交替投影神經網路，並將其應用范圍從實數域拓廣到復數域，且給出了在復數域仍然成立的若干結論.運用這些結論，在對網路噪聲抑制、網路收斂速度及待外推信號因截斷而造成頻譜嚴重外泄問題的分析與討論的基礎上，提出了一種基於交替投影神經網路的外推演算法.模擬實驗表明該方法是行之有效的.另外，該演算法對頻譜外推同樣適用；由於它採用全互連神經網路結構，易於并行計算和vlsi實現，從而可滿足軍事上實時處理的需要 -
Continuous - time filter needn ’ t a / d switch, data processing, d / a switch technique and so on to finish filtering function with analog signal like digital filter. and it needn ’ t switch action to sample data in the discrete time with analog signal like switched capacitor filter. so continuous - time filter has several advantages, such as high processing speed, simple circuit configuration, and low power consumption
對每種方法都進行了詳細的推導，總結了每種設計方法的設計步驟、適用場合，舉例用每種設計方法對具體濾波器進行了電路設計、靈敏度特性分析、失真特性分析等，並用計算機模擬驗證了電路方案。 -
( 2 ) damage detection based on wavelet transformation using the principle of time - frequency analysis based on orthonormal discrete wavelet transformation, it can detect damage by analyzing how the frequency content of observation signal changes with time
該方法利用基於離散正交小波變換的信號時頻分析原理，通過分析觀測信號的頻譜隨著時間的變化情況來檢測破損。 -
According to the special structure of the sampled - data control system, the signal channel is really composed of two parts : a continuous - time part and a discrete - time one
為此，針對采樣控制系統的結構特點，將頻率響應分為兩個通道進行計算。 -
The discrete fourier transform algorithm can analyze time - domain ' s stationary disturbance signal effectively and confirm the signal ' s distributing in the frequency domain
而離散傅立葉變換則可有效地對穩態擾動進行分析，得出時域穩態信號在頻域中的分佈情況。
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