語音編碼演算法 的英文怎麼說

中文拼音 [yǔyīnbiānmǎyǎnsuànfǎ]
語音編碼演算法 英文
speech encoding algorithm
  • 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...
  • 演 : 動詞1 （演變; 演化） develop; evolve 2 （發揮） deduce; elaborate 3 （依照程式練習或計算） drill;...
  • 算 : Ⅰ動詞1 （計算數目） calculate; reckon; compute; figure 2 （計算進去） include; count 3 （謀劃;計...
  • 法 : Ⅰ名詞1 （由國家制定或認可的行為規則的總稱） law 2 （方法; 方式） way; method; mode; means 3 （標...
  • 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. 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
  2. After about two years " insisting and hard working, this goal set at the beginning has become true. the developed c54x general assembly program for g. 729 speech signal compressing algorithm has passed the tracking with more than 3, 000 unitary standard measuring vectors. g. 729 speech signal compressing compiler using c54x general assembly program has been accomplished real - timely, and undistorted rebuilt speech signals have been obtained

    因此本課題選用c54x的通用匯編語言編程實現g . 729語音壓縮編碼演算法，調試並通過了統一標準測試矢量三千多幀，最終在5402開發實驗板上實時實現了g . 729語音壓縮編碼器，獲得未失真的重建語音信號。
  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. 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。
  5. 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
  6. The project uses for reference the algorithm thought of sbc ( subband coding ) to measure off the audio to the corresponding frequency width and encode it by the different sensitivity of human hearing, which results in the lower coding rate and bearable voice quality. the algorithm processing low bit - rate audio is designed to be self - adaptive by the situation of network. the component developped by that algorithm and project has already been used in the realtime interactive educational system

    該方案借鑒sbc （ subbandcoding ）子帶編碼演算法思想，將音頻按對人聽覺敏感程度不同劃分為相應的頻帶並進行相應的編碼，從而得到較低的編碼率和較好的語音質量，設計了可根據網路狀況進行自適應的低帶寬音頻處理演算法。
  7. Neural networks are used more frequently in lossy data coding than in general lossless data coding, because standard neural networks must be trained off - line and they are too slow to be practical. in this thesis, statistical language model based on maximum entropy and neural networks are discussed particularly. then, an arithmetic coding algorithm based on maximum entropy and neural networks are proposed in this thesis

    傳統的人工神經網路數據編碼演算法需要離線訓練且編碼速度慢，因此通常多用於專用有損編碼領域如聲音、圖像編碼等，在無損數據編碼領域應用較少，針對這種現狀，本文詳細地研究了最大熵統計語言模型和神經網路演算法各自的特點，在此基礎上提出了一種基於神經網路和最大熵原理的算術編碼方法，這是一種自適應的可在線學習的演算法，並具有精簡的網路結構。
  8. Enhanced gain filtering of g. 728 speech coding algorithm

    728語音編碼演算法的增益濾波
  9. 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
  10. Experiment results also show that our improved algorithm achieves the same perceptual quality as g. 726 standard while our improved algorithm uses a lower bit rate. therefore, our proposal may lead to telecommunication bandwidth saving and storage requirement reduction

    經實驗結果驗證，本語音編碼演算法與g . 726語音波形編碼標準相比，比特率下降了15 . 19 %以上，同時兩者的語音質量完全沒有差別。
  11. We first introduce the basic methods of speech processing in time domain. emphatically, we describe linear prediction and tonality detection of speech signal. moreover, we discuss the g. 726 speech waveform coding standard in details

    本文首先介紹了語音波形時域分析處理的基本方法，對語音波形線性預測和音調檢測技術作了重點描述，著重研究了g . 726語音波形編碼演算法，並在此基礎上，對該演算法進行了某些探討改進，並用vc + +編程，在pc機平臺上予以實現。
  12. In this paper, we investigate speech waveform coding technology with emphasis on the g. 726 recommendation of itu - t. based on g. 726, we present a new algorithm. compared with g. 726, our proposed algorithm achieves the same perceptual quality with lower bit rate

    本文結合當前商用市場對語音編碼的需求，研究了語音波形編碼技術，重點研究了itu - tg . 726建議，並在此基礎上探討了進一步降低比特率的演算法，使本語音編碼演算法的音質和g . 726演算法的完全一樣，同時，採用本文演算法的比特率低於採用g . 726語音編碼演算法的比特率。
  13. This is the background of our wi speech coding project

    本課題即是圍繞低速率wi語音編碼演算法展開的。
  14. Study on g. 729 e speech coding algorithm

    語音編碼演算法初探
  15. In this paper, we use the texas instrument corporation ' s tms320c541 chip to realize the g. 723. 1 algorithms

    723 . 1語音編碼演算法，經本系統處理后的實際合成語音具有較高的編碼質量，能達到通信的要求。
  16. Digital signal processing is widely applied for its realtime process ability. the principles and implementation of g. 729 algorithm on dsp processor are discussed in this paper

    數字信號處理因其實時快速的處理能力在通信領域得到了廣泛的應用，本文討論的是在dsp上實現g . 729語音編碼演算法。
  17. Digital speech ' s store is also an important field that requires high quality speech coding algorithm because now all kinds of portable digital recorder is more and more popular

    另外在語音存儲領域，近年來隨著各種便攜數碼錄音裝置的流行，對高合成語音質量的語音編碼演算法也提出了迫切的要求。
  18. Waveform interpolation speech coding is one of the most potential low - rate speech coding algorithms in recent years. with high performance, wi technique has been widely concerned

    波形內插（ waveforminterpolation ， wi ）語音編碼是近年來發展起來的一種非常有潛力的低速率語音編碼演算法，因其良好的性能，受到了研究人員的廣泛關注。
  19. Melp which can work at the rate of 2. 4kb / s has been chosen as u. s. federal standard, melp algorithm is on the basis of line prediction ( lp ). five auxiliary characters has been introduced into melp algorithm, they are mixed excitation, aperiodic pulse, fourier magnitudes pulse dispersion and adaptive spectral filtering

    作為一種重要的低速率語音編碼演算法，美國聯邦標準melp演算法對lpc - 10編碼方案做了大量改進，引入了混合激勵，非周期脈沖，殘差付氏幅度譜，脈沖散布和自適應譜濾波五個附加特徵，在2 . 4kbit / s的速率下取得了比較自然的語音質量。
  20. Some new methods are presented in this paper to improve the quality of harmonic excited linear predictive ( help ) coding which telecommunication and signal processing laboratory in beijing polythenic university develop

    本文以北京工業大學通信與信號處理研究室開發的2 . 4kb s諧波激勵線性預測（ help ）低速率語音編碼演算法為基礎，針對進一步提高語音質量的問題，研究了一系列改進方法。
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