code speech 中文意思是什麼

code speech 解釋
編碼語言
  • code : n 1 法典；法規。2 規則，準則；（社會、階級等的）慣例，習俗，制度。3 （電）碼，代碼，密碼，暗碼；...
  • speech : n. 1. 言語；說話；談話；說話能力（或方式）。2. 民族語言，方言，專門語言；〈罕用語〉流言。3. 演說，演講；發言。4. 【語言學】詞（類）；引語；用語。5. （樂器的）音，音色。
  1. Speech by the secretary for justice, ms elsie leung, at a conference on the bicentenary of the french civil code at the city university of hong kong on november 9, 2004

    終審法院首席法官李國能于北京的中國人民大學法學院就香港特別行政區的司法制度發表演說九月二十一日
  2. 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。
  3. Digital speech has preponderance over analog speech in reliability, robustness and security during communication. however, digital speech needs more bandwidth than the analog signal. especially with the requirement for communication frequency increasing, it ' s necessary to code speech signal at low rates

    但是，數字化后的信號所佔的頻帶大幅增加，特別是在帶寬需求日益增長的今天，這個問題尤為突出，因此語音的低速率編碼（即壓縮編碼）成為迫切的要求。
  4. Digital speech code

    數字語音編碼
  5. 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
  6. Code - switching in speech communication

    言語交際中的語碼轉換
  7. It is introduced in detail in this thesis, too. thirdly, on the basis of analyzing the capability requirement of g. 723. 1 ’ s real time realization, various methods are used to optimize the original code. finally, g. 723. 1 speech coding and decoding are realized on tms320vc5501 dsp in real time

    接著設計了整個系統的軟體結構，然後在分析g . 723 . 1標準實時實現的性能指標的基礎上，對標準給出的演算法進行了多方面的優化，最終在tms320vc5501dsp上實時實現了該標準，話音質量良好，達到了通信質量的要求。
  8. And add c source code of g. 729 to vc + + programs which i developed, so i can tell from whether the input speech is same with output

    在本人開發的vc實時語音處理的應用程序內加載g . 729協議的c語言模塊。
  9. Telephone circuit produce dtmf signal, receive ring signal and perform speech function. dtmf receiver lets it becomes binary code for mcu to read. analog circuit process speech signals between phone and gsm modem

    固話電路提供產生的雙音多頻信號及振鈴和通話功能，雙音多頻收號電路將dtmf信號變成二進制碼，便於mcu的識別和處理。
  10. The neural network for gain filter in speech code algorithm

    語音編碼演算法的神經網路增益濾波器
  11. Linear prediction code and mel - scale cepstrum coefficients are effective method in speech feature extraction and important in speech signal processing

    線性預測分析和mel倒譜分析是對語音信號特徵提取比較有效的兩種方法，在語音信號處理中有著重要的應用。
  12. Chapter 2 presents the background of speech code

    論文第一章介紹了當今語音編碼的發展。
  13. G. 729 not only has short code - decode delay but provides good synthesized speech at 8kb / s. moreover it can work reliably in many circumstances

    G . 729不僅在8kb s的速率下提供了良好的合成音質，而且理論編解碼延時較短，在多種場合均能有效工作。
  14. 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。
  15. Code deviation usually occurs in speech communication

    言語交際中往往伴隨著代碼偏離現象。
  16. Rapid development of data business, growing of packet network technology, and increasing of communication channel capacity, etc, bring this problem the answer : the next generation network will be base on the ip, and it will be to consist of network architecture which are diverse, synthetic and open such as speech sound, data, multimedia etc. the principle of voip ( voice over internet protocol ) is not complicated : at the sending end, sample the analogue speech sound signal, code and compress, then package and transmit it over the packet network

    數據業務的快速發展、分組網路技術的成熟、數據網路通信通道容量的不斷增加等給這個問題提供了答案：下一代網路將是基於ip的，下一代網路將是可以提供包括語音、數據和多媒體等各種業務的、綜合的、開放的網路構架，而voip正是這個答案的具體體現。實現voip的原理並不復雜：將模擬的語音信號采樣、編碼並進行壓縮，封裝在數據網路的分組中進行傳輸，在接收端對數據進行解碼、數模轉換恢復成模擬信號即可。
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