信源率 的英文怎麼說

中文拼音 [xìnyuánlǜ]
信源率 英文
message source rate
  • 源 : 名詞1. （水流起頭的地方） source （of a river）; fountainhead 2. （來源） source; cause 3. （姓氏） a surname
  • 率 : 率名詞（比值） rate; ratio; proportion
  • 信源 : informatio ource
  1. In all kinds of complicated network, oriented linking and unlinking, communication frequency resource is strained, and bandwith to transmitting audio frequency signal is too restricted, complicated and fluky, while audio frequency data exponential have been increased in the last several years. under the circumstances, based on the research of predecessor, this paper studies wavelet analysis ' s maths gist and practices significance on signal process, and puts forward a optimized wavelet package condensation arithmetic to process audio frequency data, which gives attention to coding efficiency, multirate and compression delay. simulation experiment on the arithmetic has been done by matlab

    針對無連接和面向連接的各種復雜網路環境下，通信頻帶資源緊張，音頻傳輸帶寬有限且復雜多變，而各種音頻數據又日益增多的局面，本文研究小波分析在信號處理方面的數學依據和在數據壓縮方面的實際意義，在前人不斷工作的基礎上，提出了一種優化小波包變換編碼方案用於音頻數據的壓縮演算法，兼考慮了編碼效率、多碼率和壓縮時延多個方面，並在matlab環境下做了模擬實驗，對各種音頻信號及多種小波函數做了模擬結果比較，實驗結果證明該演算法可以在一定計算復雜度下可以很好地改進壓縮效果，達到多碼率下實現實時編解碼的過程，在高速dsp晶元等硬體設備支持下，可以有效應用於實際復雜多變信源編碼。
  2. Simulations for high speed atm network and low speed atm network show the proposed algorithm can reduce cell loss rate and depress the fluctuation of the cell transmission rate

    對低速網路和高速網路的模擬表明了erica改進演算法大大地減少了信元損失率，降低了信源速率調整的波動。
  3. In the proposed method, the controller takes the buffer length as congestion indication, takes sources quality and bandwidth utility as object function so as to learn on line. as the controller outputs, the coding rate for input traffic sources and the corresponding user percentage are used to adjust the cells " arrival rate to the multiplexer buffer. compared with the previous method where cells " arrival rate is tuned only by the encoding rate and the encoding rates for all input traffic sources are regulated in a body, the proposed method guarantee that the quality of cells are optimal while cell loss rate is minimized, which means quality of service is guaranteed

    在該方法中，擁塞控制器以緩沖區大小信元作為擁塞指示，以信源質量和帶寬利用率作為目標函數進行在線學習，控制器輸出包括信源編碼率及其對應的用戶數在全部用戶中所佔的百分比，即根據信源編碼率及對應的用戶百分數調整信源輸入流，從而克服了以往擁塞控制方法中僅僅調整編碼率帶來的對所有信源進行整體調整的缺陷，使控制系統在信元損失率最小情況下確保信源輸入流質量最高，從而有效地利用了網路帶寬。
  4. A novel joint estimation algorithm for ranges, frequencies and doas of multiple near - field narrowband sources is proposed in the near field is proposed

    摘要提出了一種新的多個近場窄帶信源距離、頻率及到達角三維參數的聯合估計演算法。
  5. The rate - distortion function, and source coding theorem

    率失真函數和信源編碼定理。
  6. This method provides unequal protection for spiht bet stream with different importance, and adaptively adjusts the source and channel coding rates according to the time - varying characteristic of the channel, thus yields good performance and high reliability without adding extra bandwidth. simulations in rayleigh channel show that the scheme can obviously improve the image quality compared to eep and uep scheme, especially when the channel is in bad condition

    本方法通過對spiht編碼碼流重要性的不同而進行不同程度的保護，並利用通道的時變特性自適應地調整信源和通道編碼速率，從而在不增加額外帶寬的前提下有效地提高了系統的性能和可靠性，經過計算機模擬模擬，得出了在瑞利通道中，傳輸條件惡劣的情況下，本方法比eep和uep能更好的提高重建圖像的質量。
  7. In packet based wireless video communications, when best band distribution, not only source distortion should be considered, but also channel distortion caused by channel errors. the existing macroblock - layer rate control schemes calculates quantization parameters of all macroblocks ( mb ) in a frame in a raster scan order, and then encodes the mbs in the same order. actually, the quantization distortion is heavily dependent upon the coding order of mbs

    在基於包的無線視頻傳輸中，最佳帶寬分配時，不僅要考慮信源編碼失真，也要考慮由於通道差錯引起的通道失真；已有的宏塊層碼率控制演算法以矩陣掃描的順序計算一幀中所有宏塊的量化參數，事實上量化失真與宏塊的編碼順序有很大關系，改變宏塊的編碼順序，使復雜的宏塊分配到更多的比特數，顯然能大大提高編碼效率。
  8. We can use an alternative strategy to predict when congestion is about to happen, and reduce the rate at which hosts send data just before packets start being discarded

    人們對于擁塞的另外一種選擇是採取擁塞避免，即在擁塞將產生時就預測到，在分組被丟棄前降低信源的發送速率。
  9. In the proposed flow control method for abr service in atm networks, by introduction of delay factors, the cell transmission rate for abr traffic with different rtd are adjusted differently when the surplus bandwidth varies, and the rate variation amplitude for the short rtd traffic will be enlargeed while one for the long rtd traffic will be lessened, compared with erica

    改進的erica演算法加入時延因子，使在連接中具有不同時延的用戶在剩餘帶寬變化時，速率的調整具有不同反應，時延大的反應較遲鈍，時延小的反應較敏感。改進演算法使得小時延信源能更快的適應帶寬的變化，緩解了由於大時延信源速率調整的滯后而無法及時解除擁塞的現象。
  10. With the frequency become more and more destinty, the frequency divided into kinds of communication systems become more and more narrow

    通信頻率資源的日益緊張，分配到各類通信系統的頻率間隔越來越密。
  11. A testbench program is edited to simulate the behavior of the fifo. after the software simulation is accomplished, a real hardware circuit is designed to multiplex two data channels ( 1553b data channel and 1394 data channel ) according to ccsds standard. during the experiment and hardware debugging, the output logic of the fpga is checked up

    設計中，用vhdl語言對高速復接器進行行為級建模，為了驗證這個模型，首先使用軟體進行模擬，通過編寫testbench程序模擬fifo的動作特點，對程序輸入信號進行模擬，在軟體邏輯模擬取得預期結果后，繼續設計硬體電路，設計出的實際電路實現了將來自兩個不同速率的信源數據（ 1394總線數據和1553b總線數據）復接成一路符合ccsds協議的位流業務數據。
  12. In this paper we analyze the characteristics of the source of space data system and give three types of source packet model ; the relationship between m _ pdu multiplexing efficiency and transfer frame completion waiting time has been educed by the analysis of the process of packet multiplexing and frame completion ; then we analyze several key aspects that affect the protocol throughput performance metric to formulate the throughput performance metric of aos packet service

    本文在理論方面，在總結分析空間數據系統信源特徵基礎上，建立三種信源包模型;通過分析包通道復用與成幀過程，得出m _ pdu復用效率與成幀等待時間的關系;探討了三種虛擬通道復用方案;對影響aos協議包業務吞吐量性能指標的幾個關鍵因素加以分析，推導出aos協議包業務吞吐量指標計算公式。
  13. With regard to the flow regulation of the best - effort traffic, the controllable traffic in high speed computer communication networks, the present paper proposes a novel control theoretic approach that designs a proportional - integrative ( pi ) controller based on multi - rate sampling for congestion controlling. based on the traffic model of a single node and on system stability criterion, it is shown that this pi controller can regulate the source rate on the basis of the knowledge of buffer occupancy of the destination node in such a manner that the congestion - controlled network is asymptotically stable without oscillation in terms of the buffer occupancy of the destionation node ; and the steady value of queue length is consistent with the specified threshold value

    本文從控制理論的角度出發，針對計算機高速網際網路中最大服務交通流即能控交通流的調節問題提出了一種基於多速率采樣的具有比例積分（ pi ）控制器結構的擁塞控制理論和方法，在單個節點的交通流的模型基礎上，運用控制理論中的系統穩定性分析方法，討論如何利用信終端節點緩沖佔有量的比例加積分的反饋形式來調節信源節點的能控交通流的輸入速率，從而使被控網路節點的緩沖佔有量趨于穩定;同時使被控網路節點的穩定隊列長度逼近指定的門限值。
  14. In the source - channel approach, the translation probability is expressed as a language model and a translation model

    基於信源通道的方法將翻譯概率表示為一個語言模型和一個翻譯模型。
  15. In chapter 4, the purpose of this chapter is to establish a kind of strong deviation theorems of functional for the sequences of arbitrary continuous random variables, by using the conception of log likelihood ratio, and extend the strong deviation theorems on the differential entropy for dependent arbitrary continuous information sources on the the probability space (, . f, p )

    使得對于在概率空間（ ， f ， p ）上的任意連續型信源的微分熵的強偏差定理是本文的推論；第五章，總結本文的主要結論。
  16. The approach extends the parallel factor ( parafac ) analysis model from the common data - domain and subspace multiple invariance sensor array ( mi - sap ) formulations to the cumulant one, and forms three - way arrays by using the cumulant matrices got from array outputs, and analyzes the uniqueness of low - rank decomposition of the three - way arrays, then jointly estimates the ranges, frequencies and doas from the matrices via low - rank decomposition

    該演算法將通常在數據和子空間域應用的平行因子分析模型擴展至高階累積量域，利用陣元輸出計算的高階累積量矩陣構造三面陣，分析了該三面陣低秩分解的唯一性，並從分解得到的多個矩陣中聯合估計信源距離、頻率及到達角。
  17. At the same time, this thesis analyz es the optimal resourse allocation problem in video coding and reviews recent advances in rate - distortorn optimized video coding

    同時本文還分析了視頻編碼中的最佳資源分配問題，詳細討論了率失真優化理論在信源編碼和通道編碼中應用的最新進展。
  18. Atm is a typical high - speed network technology. atm networks can effectively integrate all traffic types and services into a single network, such as voice, picture, and data. the technology of integration raises the utility of network resource

    異步傳輸模式（ atm ）是一種典型的高速綜合業務網路傳輸技術，能靈活有效地統計復用具有不同服務質量要求和不同帶寬要求的不同類型的信源信號，如話音、圖像、數據等，提高了網路資源的利用率。
  19. We want to use novel switching algorithm that help us make the variable - length switching efficient. we do not need to cut the variable - length packet into fix cell as the traditional switching algorithm. in this paper, we suggest to use pps for the variable - length packet switching

    另外一方面人們對支持變長數據包的交換演算法有很大的興趣，傳統的切割變長數據包成為定長信源，在重新組合變長數據包的交換演算法降低了變長交換的效率。
  20. Since ldpc codes have a wonderful future, an adaptive coding scheme was developed based on quasi - regular ldpc codes and it encodes the source bits with different code rate by dividing the original parity check matrix properly

    由於ldpc碼在未來的廣闊應用前景，本文提出一種針對結構化ldpc碼的自適應編碼方案，對原始校驗矩陣適當變化，即可對信源信息進行不同碼率的自適應編碼。
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