語音編碼 的英文怎麼說

中文拼音 [yǔyīnbiānmǎ]
語音編碼 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
  • 編 : Ⅰ動詞1 （編織） weave; plait; braid 2 （組織; 排列） make a list; arrange in a list; organize; gr...
  • 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
  • 編碼 : encoded; code; coded; encrypt; codogram; coding編碼表 encode table; 編碼程序 builder; 編碼尺 code...
  1. Phonological activation of disyllabic compound words in the speech production of chinese

    言語產生中雙詞素詞的語音編碼
  2. Study of phonological encoding of chinese disyllabic compound words in patients with mild cognitive impairment

    輕度認知功能損害患者漢語雙詞素詞的語音編碼研究
  3. In this paper, combined with currently voice coding technique, espically with the fabulosity development of the mixed voice coding and the increasingly utility of the digital signals processor. we investigated the voice coding technique and discussed emphasizedly the technology of variable rate voice coding technology

    本文結合當前語音編碼技術尤其是混合編碼技術的驚人發展及數字信號處理器的日益實用化，研究了語音編碼技術，並重點討論了變速率語音編碼技術。論文簡要介紹語音編碼技術中的波形編碼和聲碼器的主要性能。
  4. The optimal gain filter of ld - celp

    低延遲碼激勵語音編碼演算法的最佳增益濾波器
  5. 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系列dsp晶元） ，而且其成本在不斷降低，這就為復雜的語音編碼器的實用化和產品化奠定了基礎。
  6. In this thesis, the research focuses on pitch detection techniques of the low - rate wi speech coding. aimed at the problems of voiced - unvoiced error, pitch doubling and halving, accuracy of pitch detection and pitch quantization, a series of pitch detection techniques including pre - processing, pitch detection and pitch quantization were proposed

    本文就低速率wi語音編碼中的基音檢測技術進行了深入研究，針對基音檢測中的清濁誤判、基音加倍減半、基音檢測精度及基音量化問題，提出了包括基音檢測前端處理、基音檢測演算法及基音量化的一整套基音檢測技術。
  7. It synthesizes the excellence of wave coding and parameter coding, adopts vector quantity, analyse - synthesize, perceptual weighting, therefore, gains good speech coding quality at 8kbit / s. cs - acelp can be used in individual telecom, iphone, c / n, microwave telecom and isdn

    Cs - acelp演算法綜合了波形編碼和參數編碼的優點，以自適應預測編碼技術為基礎，採用了矢量量化、合成分析和感覺加權等技術，在8kbit / s速率上獲得了較高的語音編碼質量。
  8. 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系統中，綜合語音質量和編碼速率， evrc是最佳的語音編碼器。
  9. The basic characteristics of the current data network are point - to - point, connectless, doing one ' s endeavor, no quality of service, etc. these characteristics do not meet the requirement of real - time services, therefore, the realization of voip need support of the some key technology. these technologies includes : speech sound coding and data compression, real - time transmission and control, mute compression and multicast, acoustic - echo cancellation and comfort noise generator, dynamic monitor and guarantee of quality of network service, as well as, the compatible of different network and different protocol with each other

    但現有的數據網路的基本特性：點對點的、無連接的、盡力而為的、沒有服務質量保證等特性並不適合與實時的業務要求，因此voip的實現需要一些關鍵技術的支持，這些技術包括：語音編碼和壓縮技術、實時傳輸和控制技術、組播技術、靜音壓縮和舒適噪聲生成技術、回聲消除技術、網路服務質量的動態監測和保證技術、以及不同的網路、不同的協議之間的互連互通等等。
  10. 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。
  11. In communicaton the bandwidth is an important problem that we should consider, specially in wireless communication. in fact the fiber is mainly used in backbone networks, so it is essential to develop the low rating coding technology of voice. the arithmetic of melp is based on the model of lpc and use the form of mixed excitation. because it integrates the idea of multi - band, so it has the merit of lpc and mbe. it is a perfect coding scheme in low rating voice coding relatively

    而melp語音壓縮編碼演算法是在線性預測編碼參數模型的基礎上，採用混合激勵的形式，並且結合了多帶的思想，因此它擁有線性預測編碼和多帶激勵的優點，是目前低速率語音編碼中一種比較理想的編碼方案，也是本文研究的重點。本論文通過研究melp的語音編解碼演算法的原理，對它的編解碼過程作了比較深入的研究，對其中的一些公式進行了理論推導，並作了模擬分析，最後研究了該演算法的c語言實現。
  12. Arma predictive model based celp speech coding algorithm

    基於零極點預測模型的碼激勵線性預測語音編碼演算法
  13. Voice encoders may be further enhanced by encoding digital signals at variable encoding rates

    語音編碼器可以通過在可變編碼率編碼數字信號進一步加強。
  14. Digital speech code

    數字語音編碼
  15. A new algorithm of 4kb s low bit - rate speech coding

    低速率語音編碼的一種新演算法
  16. We select fpga of type xc3s200 as hardware to design the coder and display the hardware resources inside, moreover study the method and steps of designing dsp, based on fpga, by using system generator, finally, it emphasizes the design process of multi - band excitation vocoder. we can work out the module of high pass filter and the module of low pass filter, module of divide frame, module of keynote rough estimate, module of keynote fine estimate, module of band - separated v / u judgment / verdict and module of band - separated amplitude estimate, by using simulink, ise and system generator

    本文選用型號為xc3s200的fpga作為設計編碼器的核心硬體，介紹了其內部所含的硬體資源，並研究了利用systemgenerator基於fpga設計dsp的方法和步驟，最後，本文把重點放在多帶激勵語音編碼器的設計上，利用simulink ， ise和systemgenerator分別設計其中的高通低通濾波器模塊、分幀疊加模塊、基音粗估模塊、基音精細估計模塊、分帶v / u判決模塊、分帶幅度估計模塊。
  17. 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
  18. Chosen the cvsd as the radio station ’ s voice coding. the character of strong anti - interfere in real communication circumstance was our major consideration to choose cvsd. as to validate the algorithm of cvsd and get the better parameter, we do the cvsd algorithm simulation firstly with matlab and the simulation tested the algorithm as well as given us some tips during the hardware debugging

    在實際設計和實現語音編碼過程中為了對硬體編程語言有好的指導作用以及得到比較好的編碼參數，首先用了matlab對語音編碼進行了模擬，通過模擬驗證了演算法同時獲得的參數為實際的硬體實現提供了一定的參考價值，也給我們的實際調試指出了方向性的指導。
  19. Robust speaker verification in low bit rate channels the influence of speech coding on text - independent speaker verification was studied

    研究了多種低速率通道環境下，語音編碼對與文本無關說話人確認的影響。
  20. Vector quantization ( vq ) is an important technology in the field of image compression, which is widely used in various applications such as speech coding, audio and video compression, and teleconferencing systems

    矢量量化（ vq ）是近年來圖像壓縮研究中的重要技術，廣泛應用於語音編碼、音視頻壓縮和遠程會議等系統中。
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