語系碼 的英文怎麼說

中文拼音 []
語系碼 英文
language code
  • : 語動詞[書面語] (告訴) tell; inform
  • : 系動詞(打結; 扣) tie; fasten; do up; button up
  • : Ⅰ名詞(表示數目的符號或用具) a sign or object indicating number; code Ⅱ量詞1 (指一件事或一類的...
  • 語系 : [語言學] family of languages; language family
  1. On the basis of the study on the speech coder algorithms, paper describe an advanced method of developing dsp system software, and as the guidlines, we developed the programme of whole decoder unit. paper stress on analysis of the ecu in decoder unit. aiming at amr algorithms disadvantage of angularity of synthetical speech, paper study on the specutral extrapolation which apply to extrapolate reflect coefficient of track model to make error conceal processing of amr. at last paper analyze existing echo cancellation algorithms using on mobile communication system

    在此基礎上,描述了一種較為先進的大型dsp統程序開發策略,並以此為指導思想,以美國ti公司c6000dsp開發平臺開發出了整個amr解器單元的統程序。論文對amr解器的誤隱藏處理單元進行了重點分析,針對原有演算法合成音自然度不好的缺點,論文研究了將譜外推法應用到amr演算法中外推出聲道模型反射數參數進行誤消除處理。
  2. 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「模糊譜相減法音增強」 ;針對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音端點的初始和改進參數表;提出了利用基於線性預測編倒譜參數和差分線性預測編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音識別統,在保證統實時性的同時,實現連接漢音識別統識別率的提高;在硬體實現上,詳細闡述了基於tms320vc5402的連接漢音識別統各部分硬體設計;在軟體開發上,給出了連接漢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。
  3. Tvb helped develop and introduced the nicam system to hong kong, providing digital stereo and multi - lingual programming

    開始採用麗音多聲道廣播統,為觀眾提供數式立體聲及多種言的廣播服務。
  4. Recognising that there are at least three tiers of sino - thai lexical correspondences ( old chinese, middle chinese, modern chinese ), this study makes claim on the sinitic origin of constituent morphemes in polysyllabic words in thai and explores the wider implications on sino - thai linguistic links

    漢臺有三個層次(即與上古漢、中古漢、與近代漢的關) 。本文指證近代泰有漢成份並試圖探討其由來與意義。
  5. It preestablish a parallel design pattern for each parallel project based on master / slave architecture, and the communication between computers is transparent to programmers. in another words, when programmers are developing parallel programs, they do n ' t need to learn any complicated primitives and write any codes related with communication between parallel tasks. just only set some parameters in the visual interfaces that provided by platform, and the codes will be generated by our system automatically and inserted to the fittest place in the files

    統為每一個并行計算工程設定了一個基於master slave架構的并行編程模型,並且使得各結點機之間的通信對于開發人員透明化,也就是說,開發人員在開發并行程序時,無需了解紛繁復雜的通信原,不涉及任務之間通信的代,只需要在并行程序開發平臺提供的可視化界面上對數據的傳送進行設定,通信代將自動生成並插入到程序合適的地方。
  6. With the developing of vlsi in recent years, high function dsp has been produced ( such as tms320 series dsp produced by ti ) and their cost is dropping. thus, this established the foundation for making complex speech coder practical and producible. the paper researched and discussed the fix - point real implementation of g. 728 by dsp tms320c5402 chip

    但是,近幾年來,隨著大規模集成電路( vlsi )的發展,已生產出高性能數字信號處理晶元(例如ti的tms320列dsp晶元) ,而且其成本在不斷降低,這就為復雜的音編器的實用化和產品化奠定了基礎。
  7. Jfs logging semantics are such that, when a file system operation involving meta - data changes - that is, unlink - returns a successful return code, the effects of the operation have been committed to the file system and will be seen even if the system crashes

    Jfs日誌記錄的義如下:當涉及元數據更改的文件統操作- -例如, unlink ( ) - -返回成功執行的返回時,操作的結果已經提交到文件統,即使統崩潰了也可以發現。
  8. It is expected to be used for 3g personal handy - phone system as standard algorithms which encode speech signals and decode it. additionally, this kind of algorithms which own excellent quality can be application in viewphone and video order programme etc. the thesis introducethe algorithm structure of g. 729

    該協議在可預見的將來可能應用於三代移動通信統中作為音編解演算法。另外,由於其良好的性能也可應用在多媒體統中如:可視電話,視頻點播等。本論文概要介紹了g . 729協議的演算法結構。
  9. At the present time, evrc is the best vocoder in the cdma system when take into account both the voice quality and the encode rate

    在目前的cdma統中,綜合音質量和編速率, evrc是最佳的音編器。
  10. The protocol specifies the definition of inter - library loan service working on c / s mode. by means of 21 apdu messages defined in this protocol and standard encode / decode rule, we can realize the resource sharing between different libraries. this article advances a set of solution scheme for the implementation of inter - library loan on the internet environment, and gives some discussion about the key elements for the implementation which includes the process of apdu, the computer descriptions of asn. l syntax, the ber coding rules and the design for supporting db etc. the exact method and program for implementation of inter - library loan are also presented in the article

    本文在詳細分析和研究館際互借協議內容和工作原理基礎上,根據我國圖書館目前的實際情況,給出了在internet環境下實現館際互借服務的解決方案,論述了基於windows2000操作統下開發館際互借軟體的具體思路,討論了實現該協議的一些關鍵問題,包括apdu的處理, asn . 1抽象法表示、 ber編規則、數據庫設計等,並給出了具體的解決方法和實現程序。
  11. Second, the international standards for video coding are briefly described including h. 26x and mpeg. particularly, the encoding and decoding algorithm and bitstream syntax of h. 263 are introduced in detail

    第二:總結了針對不同應用而制定的視頻編國際標準包括h . 26x和mpeg列,較詳細的介紹了h . 263編解演算法和法結構。
  12. According to the project of adaptive multi - rate speech coding ( amr ) being put forward by the third generation group of the mobile communication, this paper takes the principle of the speech arithmetic as the base, studies the technologies including the source controlled rate, voice activity detector, comfort noise and the error concealment unit in amr, discusses its the characteristic of adaptation and analyses its performances particularly. amr c codes are researched carefully through the modules being divided into and debugged under the tms320c54x provided by the ti corporation, and optimized in selecting the method of c code embedded assembler codes and simplified in the search codebook combining with the theory of speech coding, which are based on the realization about theory and practice of the optimization of amr speech coding

    從自適應多音編演算法的c代出發,對它進行模塊劃分後作了統分析,將其在vc下調試通過,進一步在ti公司提供的tms320c54x環境中進行調試,結合音編理論,對演算法進行優化,採用了在c代中嵌入匯編和簡化自適應本和固定本搜索的方法,部分地提高了c代效率,為實現自適應多音編的優化奠定了理論和實踐基礎。
  13. A series of statements surrounded by curly braces form a block of code

    由大括號括起來的一句構成代塊。
  14. Can you tell me the number of the english department

    請告訴我英的電話號好嗎?
  15. Could you tell me the number of the english department

    請告訴我英的電話號可以嗎
  16. Utf - 8 encoding default for new systems the installer will set up systems to use utf - 8 encoding rather than the old language - specific encodings like iso - 8859 - 1, euc - jp or koi - 8

    安裝程式將會把統設定為使用utf - 8編,而不是使用專用於各的舊特定編像是iso - 8859 - 1 euc - jp或koi - 8 。
  17. Actual automatic differentiation technique can n ' t fulfil the needs of the application fields well, such as the incompatible problem in the different operating system and the nonsupporting problem with the language needed to be processed automatic differentiation transform. we made some amelioration based on the deep research on the source code of odyssee to make it suit our present appliance, such as the supporting with fortran90

    目前的自動微分技術不能充分滿足應用領域的需求,如在不同體結構下的兼容問題、對需要微分的言的不支持等,本文在深入研究odyssee源代的基礎上對其進行了改進,使之能夠符合當前應用的需要,如對fortran90的支持。
  18. The content of this paper is arranged as foll owing : chapter 1 introduces the concept of credit, credit risk and credit assessment, as well as the history and development of credit assessment ; chapter 2 introduces the history of ai technology, and the background of expert system and neural network. characters and disadvantages of expert system and neural network are presented respectively and the necessity of combining expert system and neural network is lightened ; chapter 3 shows the process of dealing with sample data, including the treatment of exceptional data and factor analysis, and puts forward the concrete framework of the mixed - expert credit assessment system ; chapter 4 introduces concept of object - oriented technology, and constructs object model and functional model after analyzing the whole system. it also illustrates the implementation of concrete classes by an example of rule class and the inference algorithm in the form of pseudocode ; chapter 5 introduces the structure of the whole system, the major functional models and their interfaces, and the characteristic of the system is also generalized ; chapter 6 summarizes the whole work, and points out the remaining deficiencies as well as the prospective of this method

    本文具體內容安排如下:第一章介紹了信用、信用風險、信用評價的概念,回顧了信用評價的歷史、發展和現狀,並綜合各種信用評價模型,指出這些模型各自的優缺點:第二章簡單描述了人工智慧技術,著重介紹有關專家統與神經網路的基礎知識,通過總結它們的優缺點,指出結合專家統與神經網路構造混合型專家統的必要性;本章還介紹了神經網路子模塊的概念,提出了混合型專家統的一般框架與設計步驟:第三章對樣本數據進行處理,包括異常數據的剔除、因子分析等,提出了信用評價混合型專家統的具體框架結構,介紹了統知識庫的主要部分、基於優先級的正向推理機制的流程、以及基於事實的自動解釋機制的具體實現方法;第四章介紹了面向對象技術,進而採用面向對象對信用評價統進行分析,建立了對象模型和功能模型,並在此基礎上,採用c + +言以規則類為例說明統中具體類的實現,用偽代的形式描述了推理的演算法;第五章描述了整個統的結構,對統主要功能模塊和界面進行了介紹,並總結統的特點;第六章總結了全文,指出本文所構造統存在的不足以及對將來的展望。
  19. At the receiving end, a inverse process is performed. the system receives low rate data and the fpga reorganizes a frame of data which is decoded by the compression chip every 20 ms. the obtained pcm signal is performed d / a to restore the analog speech signal

    在收端進行相反的過程,接收低率數據,並由fpga重新組幀,送至主晶元解得到pcm信號,再作d / a變換,恢復出模擬音,統是全雙工的。
  20. Mandarin digit speech recognition system based on dsp technology

    技術的漢音識別
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