語音代碼 的英文怎麼說

中文拼音 [yǔyīndàimǎ]
語音代碼 英文
phonetic code
  • 語 : 語動詞[書面語] （告訴） tell; inform
  • 音 : 名詞1. （聲音） sound 2. （消息） news; tidings 3. [物理學] （音質） tone 4. （姓氏） a surname
  • 代 : Ⅰ動詞1 （代替） take the place of; be in place of 2 （代理） act on behalf of; acting Ⅱ名詞1 （歷...
  • 碼 : 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...
  • 語音 : speech sounds; pronunciation; voice
  1. 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系統中作為語音編解碼演算法。另外，由於其良好的性能也可應用在多媒體系統中如：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。
  2. Section ii describes the design approach and implementation of speech module on mcf5249 coldfire core. the speech codec optimizes g. 729a codes and added voice activity detection of g. 729b to save bandwidth ; the implementation of acoustic echo cancellation uses nlms algorithm and it can reduce echo though designing adaptive fir filter and speech detector ; the dtmf and cpt generate signal using two second order digital sinusoidal oscillators and detect signal by picking up the frequency information. but only get the frequency information is not enough in cpt detector, this thesis introduces a method

    其中對語音編解碼器的設計採用優化g . 729a代碼達到設計要求，並在此基礎上加入g . 729b的靜音檢測模塊，以進一步降低網路傳輸帶寬;對回聲消除器的設計採用nlms演算法，通過設計自適應fir濾波器和語音檢測器達到回聲消除目的;對雙音多頻設計，信號發生端採用構造靜態參數表並通過二階正弦振蕩器產生信號，信號檢測端提取頻率信息以檢測信號;對呼叫進程音設計，除了類似雙音多頻的信號發生及頻率檢測設計外，還需要檢測信號持續時間，作者設計了一種基於匹配狀態表的方法以檢測信號持續時間。
  3. 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

    從自適應多碼率語音編碼演算法的c代碼出發，對它進行模塊劃分後作了系統分析，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，結合語音編碼理論，對演算法進行優化，採用了在c代碼中嵌入匯編和簡化自適應碼本和固定碼本搜索的方法，部分地提高了c代碼效率，為實現自適應多碼率語音編碼的優化奠定了理論和實踐基礎。
  4. Then, we build a common “ identity ” : a tone, an atmosphere, a style that will permeate all the facets of any future m & m products from the global concepts ( dramatic themes, general mood, etc. ) to the more specific details ( graphic codes and symbols, musical leitmotivs, etc. )

    然後，我們要創造一個通用的「特性」 ：語氣、氛圍、風格等等，以滲透到魔法門系列產品的每一個方面，整體細節包括了圖形代碼和符號、音樂主題等等。
  5. The spoken representation of a dash in morse code

    長劃莫爾斯電碼中，以短斷音代表口語表達
  6. Voice prompt will instruct you to enter your password followed by the command. enter your old password followed by the command 3 and the new password followed by “ # ” key

    根據語音提示，輸入您的密碼和功能指令。然後順次輸入原密碼，指令代碼3和新密碼。最後按#鍵結束。
  7. Both of the optimized methods for lsp parameters and algebraic codebooks can provide the similar performance as the standard method with less search load

    對分裂矢量量化碼本搜索方式的優化和代數碼本搜索方式的優化都在不明顯降低合成語音質量的情況下有效地減少了運算量。
  8. The voicexml code generated by the callflow builder is sufficient for the creation of simple voice applications, demos, and prototypes

    由callflow builder生成的voicexml代碼已經足以創建簡單的語音應用程序、演示版以及原型了。
  9. Coding of speech at 8 kbit s using conjugate - structure algebraic - code - excited linear - predictive cs - acelp

    使用共軛結構代數碼激勵線性預測的8kbit s語音編碼
  10. After phase 4, inspection and validation, the training center pi should coordinate with the corresponding air carrier pi if applicable, make an evaluation agreement on the training center regulation compliance, and submit this evaluation agreement to the regional administration flight standards office and the principal manager of the regional administration, make the decision of whether to issue the certificate or not, at the same time, apply for the certificate number of ccar - 142 training center to the flight standards department, the two character code of pinyin represent each administration, for example, hd represents east china regional administration, hb represents north china regional administration etc

    完成第四階段驗證檢查后，訓練中心主任監察員應當與相關航空承運人主任監察員進行協調如適用，就訓練中心規章符合性問題達成一致意見，並將此意見提交給地區管理局飛行標準處和管理局主管領導，做出是否頒證的決定，同時可以向飛行標準司申請ccar 142部訓練中心合格證代碼，其中漢語拼音的兩字代碼分別表示不同管理局，如hd代表華東管理局hb代表華北管理局等。
  11. 7. enter after the voice help of the computer - 0000 inquire the total bet points and result of the agent

    7依照計算機語音的提示輸入代號0000查詢代理商總押碼點數及輸贏。
  12. Fpga has become the best selection in the design of complex digital system in modern design process of digital system, especially in communication system, because of its merits like high integration, better reliability, short design period, less investment and agility. the usage efficiency of fpga for communication system can be raised by designing low rate speech coder in fpga

    在現代數字系統設計中， fpga因為高集成度，高可靠性，設計周期短和投資小逐步成為復雜數字系統設計的理想首選，尤其是在通信系統中大量地使用，把低速率的語音編碼器在fpga中設計，可以提高通信系統中的fpga的利用率，節約成本。
  13. With the booming of the third generation mobile telecommunication, the variable rate speech coding technology, which is the core of it, has been widely studied during recent years

    隨著第三代移動通信系統的迅猛發展，作為其話音業務的核心技術，變速率語音編碼技術在近年來得到了廣泛的研究。
  14. Tetra - the new digital trunked communication system, specifies algebraic codebook excited linear prediction ( acelp ), which is improved codebook excited linear prediction ( celp ), as its speech codec. the bit rate of it is 4. 567kbit / s

    本文首先詳細介紹了新一代數字集群通信系統tetra的語音編解碼採用的編碼速率4 . 567kbit s的代數碼本激勵線性預測（ acelp ）演算法。
  15. It solves payment issue through sharing pos and brush card. it solves info sharing and exchanging problem by enterprise application integration it adopts research method of software engineering and uses touch - screen, network, database technology and so on to carry through total design of the system and build the software : it uses user status identify and responsibility control to ensure database and application program ' s security ; it strengthen the code by coding optimize ; it captures and discards application runtime error to enhance the system ' s stability ; it uses multimedia voice and moving picture to show help information, thus makes the system easy to use ; it greatly reduces the maintenance work of the system by self - updating function ; it is an opening system by using star - model network top structure, supporting standard network communication protocol ? tcp / ip and offering standard software interfaced criterion

    論文採用軟體工程的研究方法，使用觸摸屏、網路、數據庫等技術，進行了系統總體方案設計和軟體開發：通過對數據庫和應用程序的用戶身份識別和權限控制，保證數據存取和應用程序的安全性；通過對代碼進行優化提高了代碼的健壯性；通過捕捉並拋出系統運行時的異常錯誤提高了系統的穩定性；通過多媒體語音、圖形和動畫提示幫助信息來增強系統的易用性；客戶端程序自動升級功能提高了系統的可維護性，有效地減少了維護工作量；系統採用星型的網路拓撲結構，支持標準的tcp ip網路通訊協議和規范的軟體介面標準，具有良好的開放性。
  16. Based on the analysis of current situation and the development trend of ip phone, this paper has put forward the solution of a kind of new - type embedded ip phone terminal and has finished designing the ip phone codec which is the key part of ip phone terminal

    針對第三代ip電話的技術現狀和發展趨勢，本論文提出了一種新型的嵌入式ip電話終端解決方案，並完成了該終端的核心部分，語音編解碼器的設計與實現。
  17. Tdma third generation wireless - adaptive multi - rate speech codec minimum performance requirements

    時分多址聯接第三代無線電通信.自適應多速率語音編碼譯碼器最低性能要求
  18. It adapts to the cdma system and achieves multi - rate speech coding and decoding. source and mode control are combines in smv for rate selection, so it improves the flexibility of cdma system, it will allow cdma subscribers to enjoy superior quality while allowing service providers to increase capacity as needed. smv is regarded as a breakthrough technology that provides significant capacity and quality gains on cdma systems, so the researching of smv is of great practical value

    可選模式聲碼器（ smv ? selectablemodevocoder ）是3gpp2最新的用於寬帶擴頻cdma通信系統的變速率語音編碼標準，它實現了語音的多種低速編碼和解碼，在速率選擇上將源控和模式控制相結合，提高了cdma系統的靈活性，可以在保證高質量語音的同時盡可能增加系統的容量，被認為是變速率語音編碼在cdma系統中應用的「突破性」技術，代表了當前語音編碼發展的方向和潮流，因此smv的研究具有很大的價值。
  19. Conjugate structure algebraic code excited linear prediction was approved as itu recommendation in 1996 based on the project of usa at & t, japan ntt, franc telecom and canada sherbrooke university. cs - acelp based on adapt linear prediction is one of the most sophisticate algorithms in the field of low bit rate speech coding

    共軛結構代數碼激勵線性預測語音編碼（ conjugatestructurealgebraiccodeexcitedlinearprediction簡稱cs - acelp ）演算法是1996年國際電信聯盟（ itu ）根據美國at & t 、日本ntt 、法國電信和加拿大sherbrooke大學聯合提出的方案而制定的，它是最復雜低數碼率語音壓縮演算法之一。
  20. To meet the needs of the third mobile communication system for higher speech quality and greater system capacity, the cdma systems employ variable rate speech coding technologies, which select the encode rate dynamically according to the signal energy and background noise. this method insures the speech quality as well as reduces, the average encode rate, so the system capacity is increased

    為了滿足第三代移動通信系統對高質量語音和大容量系統的需要，在cdma系統中普遍採用了變速率語音編碼技術，根據話音的信號能量和背景噪聲動態地決定編碼速率，這種方法既保證了話音的質量，又降低了平均編碼速率，從而增加了系統容量。
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