result packet 中文意思是什麼
result packet
解釋
結果包-
This thesis studies the law of affecting de - noise result and the selection of the threshold and the wavelet function, the combination of wavelet and fft in the fault diagnosis of turbine - generator sets : by the de - noise anslysis of blocks and sin signals, concludes : to blocks signals, usually adopts soft threshold ; the law of affecting de - noise result is when use wavelet auto - de - noise, with the increasing of decomposed level, the de - noise result becomes worse while the level blow the 3, when the level above 3 and when uses wavelet packet, it is the other way round ; the best de - noise methods of the signal is that uses " dbl " wavelet function, three level, soft and " rigrsure " threshold
本文研究了分解層數對消噪結果影響的規律和閾值、小波函數的選取，結合小波分析與fft分析診斷汽輪發電機組的故障。通過對brocks和sin兩信號的分析，得出：對blocks信號進行分析一般採用軟閾值；分解層數對消噪結果影響的規律為用小波自動降噪在分解層數小於3時，隨著分解層數的增加，消噪結果變好，反之，則變差，用小波包降噪時隨著分解層數的增加，消噪效果變好；適宜選用dbl小波軟rigrsure閾值自動消噪。 -
This paper is introduced bp neutral network character, algorithm, designing principal on its construction and the designed product. input features extracted by the way described above into a bp neutral network, and using it to classify seven type of different carburized layer depth specimen. the result is indicated, using wavelet packet method to extract features and bp neutral network to classifying, is effective and precise to classify different metal carburized layer depth. it is useful and economical
本文介紹了bp神經網路的特點、演算法和其結構的具體設計方法和設計結果，並將小波包提取的特徵值輸入到bp網路，對7種不同滲碳層深度的試件進行分類，實驗結果表明，小波特徵值提取和bp神經網路分類器相結合，可以實現對不同滲碳層深度的分類，效果良好，精度較高，有一定的實用價值。 -
In the meanwhile, we build a complete simulation model of layered wireless self - organizing routing network and verify feasibility of network architecture and key technologies, including operating mode of wireless interface, addressing and routing in lwsrn we study the performance of wsrn in terms of routing overhead, packet delivery ratio, and the communication capability, and compare these result with that of ad hoc network
同時，構造了完整的分層結構的無線自組織路由網路模擬模型。驗證了網路體系結構和關鍵技術的可行性，包括無線通道工作方式、網路編址技術、網路路由過程。並通過模擬分析了分層結構的無線自組織路由網路的路由負載、網路數據到達率和網路通信容量。 -
For packet loss result from network congestion, rab adopts the same recovery mechanisms as the original tcp ; for packet loss result from link error, it adopts a recovery mechanism which is called immediate recovery, and the value of congestion window of the source is set to a value that corresponds to the available bandwidth
對于擁塞導致丟包，則採取與傳統tcp協議相同的恢復機制（即窗口減半，並進入擁塞避免階段的線性增加） ，而對于鏈路錯誤導致丟包，則採取立即恢復策略，將tcp源端的擁塞窗口設置為與其估計帶寬相應的數值。 -
Having analyzed telegram message ( communicated between the client end and the server ) format and studied c / s exchanging the information content that mainly is two big categories : command and command executing result. this article classified the message and designed transmission data packet telegram message protocol
對客戶端與服務器之間在數據通訊時使用的電文格式進行了分析，研究了c s （ client server ）互通信息的內容，主要是兩大類：命令與命令執行的結果；本文將之歸類，設計了客戶端與服務器傳輸數據包時使用的電文協議。 -
Subsequently, taking into consideration the characteristics of audio data over internet including delay, jitter, packet loss and etc., we propose a series of methods for solving this above problems, such as pre - storage technology, buffer technology, dynamic adjustment of the voice - coding rate to the state of network and integrated media synchronization playing mechanism, and etc. in the end, simulation on 10 / 100m lan is made using the above methods, and the result of the experiment demonstrates the method has good performance and can improve the quality of the audio data transmission
其次本文還深入研究了語音數據在非實時的internet數據網上的傳輸特性，這些特性包括延時、延時抖動、數據包丟失等。在本文的設計方案中提出了針對這些問題的解決方法，包括預取機制、設置緩沖區技術、動態速率調節技術以及媒體綜合同步播放機制等。最後採用這些方法在10 100m局域網上做了模擬實驗，實驗結果表明本文提出的方法是有效的，在網路狀況惡劣的情況下能夠改善語音播放質量。 -
It was the result from that the width of spectrum of the input em pulse packet was too wide, the spectrum included rather strong frequencies being above the cut - off frequency which have not been decayed
分析認為，這是由於輸入電磁波包脈沖的頻譜太寬，存在較強的高於截止頻率的頻率，這部分頻率未被衰減而直接傳輸所致。 -
And then this thesis designs the reliable data transfer service from three aspects : packet format, the creation and free of the connection, the sending and receiving of the data. the result of the designation is the connection - oriented sending su and the connection - oriented receiving su. they work together to ensure the function of reliable data transfer
然後從包格式，建立和撤銷連接以及發送和接收數據三個方面進行設計，主要的設計結果是有連接發送服務元和有連接接收服務元，它們配合起來共同完成可靠數據傳輸的功能，並在最後給出了設計的流程圖。 -
In the simulation built up by opnet, the results show that short service time cell selection is better than the traditional method only based on sir and la - css / la - ho proposed by a. sang on the whole, making good use of spare resources and reducing the packet delay. another result shows that with condition b, ongoing call blocking probability is reduced at the cost of higher new call blocking probability, but the total blocking probability is reduced while the qos does n ' t degrade
經過opnettm編寫的多小區hsdpa模擬平臺的模擬模擬，模擬結果反映最短服務時間小區選擇的總體性能優于「基於sir的服務小區選擇」和由a . sang提出的la - css / la - ho ，提高系統的利用率，減低分組時延;接納條件b降低服務中呼叫的阻塞率，是以提高新呼叫阻塞率為代價，但在保證qos的前提下降低了總的呼叫阻塞率。 -
The simulation result shows that using wavelet packet analysis to separate the character from the transient fault signals is an efficient way
模擬結果表明：利用小波包分析理論對電力系統故障暫態信號特徵提取處理是行之有效的方法。 -
Simultaneously used the massive experimental result, we determine important parameters of detection traffic as packets sending gap and packet type
同時利用大量實驗數據確定了探測包的發包間隔、探測包模型和測量寬度等重要參數。 -
We chose suitable tcp throughput model to estimate the available bandwidth correctly, using the estimated round trip time and packet loss ratio for the next time interval as parameters of the model to achive the accuracy of estimated network bandwidth. as the observed losses and round trip time vary very dynamically, adjust the sending rate equivalent to the amount of tcp throughput may result in a rather fluctuant sending rate. so we present a rate adjustment like tcp congestion control based on aimd, which increases its sending rate by an additive inereease rate
根據mpeg4視頻流應用的特點，選擇合適的吞吐量模型，進行合理的參數估計，並根據計算出的帶寬進行相應的速率調整來實現擁塞控制，我們使用未來rtt的估計值和分組丟失率的估計值作為吞吐量模型的參數，增強了控制的實時性，弱化了業務的振蕩性，提高了帶寬預測的準確性；在進行速率調整時，不是簡單地將發送速率調整到與tcp吞吐量模型一致，而是採用類似tcp的aimd策略來調節發送速率，減小了發送速率的振蕩性。 -
As a result, it is no need for the endpoints to maintenance any system control task. in this article we propose a compactness coupling conference system architecture, and at the same time we analysis the system key techniques, giving a part of flow chart and pseudocode as examples. main content are given as below : how to use direcshow technique to obtain multimedia data ; multimedia data transport over ip network ; multicast principal and application in the system ; packing and unpacking rtp / rtcp packet so as to control network flow and in this part we introduce spillage arithmetic ; as the core part of the system, conference control and management are the main concern in our design
323框架基礎上提出緊耦合式會議系統體系結構，並以c s結構作為實現模型；本文對會議系統的中心mcu流程進行分析設計，並建立數據結構；該部分還涉及到對各個資源的調度；為了對會議控制有更好的理解，用petri網描述了會議控制會議管理的過程並提出了靈活的授權管理機制；對服務器端的數據通信設計了多播通信的實現方案；本文簡單回顧了directshow技術的應用，並在此基礎上實現了多媒體數據的採集和回放；為實現多媒體在網路上實時傳輸選擇不可靠傳輸協議udp ，為使數據能正確回放，採用了rtp和rtcp協議。 -
3. a cam - based high - speed policy engine has been successfully designed and developed using fpga, result of simulation and synthesis indicates it is able to execute high - speed packet classification preferably under oc - 48 optical network circumstance. 4
採用fpga晶元設計開發了基於cam的協議引擎原型，模擬、綜合后的結果表明它能滿足oc - 48同步光網路對輸入分組進行快速分類的要求。 -
When the pulse width of input gaussian wave packet was reduced and the length of forbidden region retained, the result displayed the output signal distorted seriously and its spectrum changed very hard. a frequency above the cut - off frequency became the main frequency and the group velocity was below than c
在保持截止區長度不變，而減小輸入高斯波包的脈沖寬度時，模擬顯示輸出波形嚴重失真，而且，頻譜也發生很大的變化，主頻已變為一高於截止頻率的頻率，群速度小於光速c 。 -
This paper is described wavelet transform theory, mother wavelet choice, the method to filter signal by wavelet transform and the result, prospered a way to extract feature originated from wavelet theory, which using wavelet packet analyzing method to subdivide signal both in low frequency and high frequency field, and consider energy of every layer as feature in frequency field, and in conjunction with the detailed analyzing character of wavelet packet in time - frequency plane, consider several minimum or maximum points in the lowest frequency band a s features in the time field
本文介紹了小波變換的理論、基小波的選擇和利用小波變換進行信號濾波的方法和濾波處理結果，並提出了一種基於小波理論的新的特徵值提取方法。即利用小波包分析方法將信號在低頻、高頻段作進一步的細分，以各層分解的能量作為信號的頻域特徵值，以最低頻帶的極值點作為時域特徵值，這樣的特徵值選取方法較全面的反映了信號的時-頻特徵，優于傳統的傅里葉分析方法。 -
3. based on the theory of analysis methods which include wavelet, wavelet packet and feedback adaptive kalman filtering, it is put forward a proposition combined with wavelet packet and feedback adaptive kalman filtering, in order to result comparative smooth to the signal tested when it is being relatively steady, and extreme strong traceability goal to the signal tested when it is taking place sudden change
在分析動態小波濾波、小波包濾波和誤差反饋式自適應卡爾曼濾波方法的基礎上，提出基於小波分析反饋式自適應卡爾曼濾波方法，以實現當測定信號相對平穩時濾波結果較為平滑，而在測定信號發生突變時濾波器又具有極強跟蹤能力的目標。 -
Wavelet transform is a wonderful method, which have adjustable resolution in time and frequency fields leading to more subtly analysis for a signal segment. it works in a good pattern according with the rule of human ears distinguishing frequencies from voice. for stft having inevitable disadvantages in analysis of unstable signal such as speech signal, as a result of study on wavelet theory and speaker recognition techniques, two feature parameters, iwptc ( incomplete wavelet packet transform coefficients ) and wptc ( wavelet packet transform coefficients ), are got based on wavelet transform
針對短時傅立葉分析在提取說話人特徵參數時的缺陷，本文通過對小波理論和說話人識別技術的研究，借鑒了一種傳統的基於聽覺機理的特徵參數mfcc （ mel頻域倒譜系數） ，利用小波變換、小波多分辨分析和小波包變換，構造出了兩種基於小波變換的說話人識別特徵參數： iwptc （不完全小波包變換系數）和wptc （小波包變換系數） 。
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