speech coder 中文意思是什麼

speech coder 解釋
語言編碼器
  • 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。
  • coder : 編碼器；編碼員
  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系統程序開發策略，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解碼器單元的系統程序。論文對amr解碼器的誤碼隱藏處理單元進行了重點分析，針對原有演算法合成語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系數參數進行誤碼消除處理。
  2. 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。
  3. In order to obtain a high - quality speech codec, the phase information of speech should be included in codec. in this thesis, a method for quantizing the phase of sew ( slowly evolving waveform ) and reconstructing sew ’ s phase with cubic polynomial interpolation is given based on the perceptual weighting analysis - by - synthesis ( a - b - s ) vector quantizer for the phase spectrum in wi coder

    本文基於感覺加權相位譜分析合成（ abs - analysis - by - synthesis ）矢量量化方法，給出了一種wi編碼器中慢漸變波形（ sew - slowlyevolvingwaveform ）的相位信息量化及合成端相位的三次多項式插值重建方法。
  4. Real time implementation of g. 723. 1 speech coder based on tm

    1語音編解碼器的實時實現
  5. The adaptation processing includes linear prediction coefficient adaptation and adaptation of quantization step size for residual signals. based on g. 726, we adopt a huffman coder to make use of probability statistic of bit cascade covering n ( n 1 ) samples generated from adpcm, in order to further reduce the bit rate. ng is lossless entropy coding, the speech quality of our improved algorithm should be same as that of g. 726 standard

    我們的研究和改進工作包括:研究最優非均勻自適應量化器，及其自適應演算法;研究波形預測函數，以及函數零點、極點的自適應演算法;基於每n （ n 1 ）個樣本所對應符號的概率統計，對預測殘差量化值再進行huffman編碼，進一步降低比特率。
  6. It is convenient to implement speech coder and decoder on a single dsps chip

    在單片dsp處理器上實現語音信號編解碼是十分便利的。
  7. 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。
  8. In this paper, we give a detail discussion on the key technology, including software and hardware designing of g. 729a multi - channel speech codec realtime implemention on a simple dsp processor - tms320c6202. in combination with the requirements of a military communication network, the atm adaptation solution of g. 729 coder bit stream is analyzed. a kind of new atm adaptation technology - aal2 is introduced. the analyse and research of aal2 are provided

    本文詳細討論了多路g . 729a語音編解碼器在一片dsp處理器tms320c6202上實時實現的軟硬體設計和關鍵技術。結合某軍事通信網設備的需要，進而對g . 729語音編碼的碼流的atm適配方案進行了分析。提出了用一種新的atm適配技術- - aal2進行適配的方案。
  9. The third part in this paper gives g. 729 compressing algorithm. the kernel of this paper occurs in the fourth chapter, namely the realization of g. 729 speech signal coder. the last part of this paper draws the conclusion and conceives this task ' s betterments

    第三章介紹g . 729的壓縮原理及演算法結構，第四章是本論文的核心， g . 729壓縮演算法在ti的dsp晶元上的實時實現，包括語音壓縮演算法實現流程及在實際工作中的一些心得體會。
  10. Itu - t has also put forward a series of ones in low rate speech coder

    ） u ? t也在此基礎之上提出了一系列關于低速率語音編碼的解決方案， g . 723 . 1標準就是其中之一。
  11. Waveform interpolation as a great potential speech coder has got much attention

    波形內插（ wi ? ? waveforminterpolation ）作為一種極具潛力的語音編碼方法受到了人們的關注。
  12. The speech coder has two kinds of bit rate, which are 5. 3kbps and 6. 3kbps, so that it can switches at frame boundaries

    這種聲碼器具備兩種比特率: 5 . 3kbps 、 6 . 3kbps 。在幀邊界處可以在兩種速率間進行切換。
  13. Cs - acelp speech coder use a special codebook structure to simplify the search procedure of codebook, has the advantage of low bit - rate, low complexity, and high speech quality

    它採用了特殊的碼本結構，簡化了碼本的搜索過程，具有低延遲，低計算復雜度和高語音質量的優點。
  14. In the study of speech coding in present, the mixed excitation linear prediction ( melp ) is a kind of relatively good method. the melp vocoder is the new federal standard speech coder

    在現有的語音編碼研究中，混合激勵線性預測編碼（ melp ）是一種比較好的方法， melp編碼方法已經被確定為美國新的聯邦語音編碼標準。
  15. Amr speech coder is a new speech coder recommendation produced by 3 gpp using on the 3rd generation mobile communication system w - cdma. this paper mainly deal with the algorithms of amr speech coder

    Amr語音編解碼標準是3gpp提出的第三代移動通信系統w - cdma的語音編解碼標準。本文主要對amr語音編解碼演算法進行了分析和研究。
  16. The goal of this thesis is to develop a kind of 2kbps wi speech coder based on waveform interpolation ( wi ) coding at 3. 75 kbps, and have it simulated on computer by c language. in this thesis, the existing wi model has been improved

    本文的主要目標是在現有的3 . 75kbps波形內插（ wi ? waveforminterpolation ）演算法的基礎上，開發一個速率為2kbps的wi編碼器，並用c語言在計算機上模擬實現。
  17. Informal subjective a / b listening tests indicated that the reconstructed speech quality of the improved wi codec with unquantized parameters is a little better than that of itu g. 726 32kbps adpcm coder

    非正式的a b測試表明，改進的wi演算法的模型質量好於itu32kbpsg . 726標準。
  18. At first, from the discrete digital model of the speech generation, the basic of speech encoding technology is briefly introduced in this thesis. secondly, some key techniques in the linear predictive speech coder of g. 723. 1 arithmetic are discussed in detail

    本文首先從語音產生的離散數字模型出發，簡要敘述了語音編碼的技術基礎，詳細討論了g . 723 . 1標準的線性預測編碼聲碼器的一些關鍵技術。
  19. H. 323 is an itu standard which provides several services including speech, data and multimedia etc. being a speech compression coder protocol supported by h. 323, g. 729 has the advantages of low bit - rate and high speech quality and been selected by itu - t as 8kb / s standard

    G . 729做為h . 323支持的語音壓縮編碼協議，具有低延遲，高語音質量的優點，被itu確定為8kb / s語音編碼標準。
  20. This system uses the evm of dual - core tms320vc5471. for the different core configuration, the dsp has a / d and d / a modules to implement speech signal i / o channel, real time implementation of speech coder. meanwhile embedded operation system uclinux is transplanted into it

    結合cpu兩個內核不同的體系結構，在dsp內核上連接a / d 、 d / a模塊實現語音信號i / o通道以及實時實現語音編解碼演算法，在arm內核上移植了clinux嵌入式操作系統。
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