speech coding 中文意思是什麼

speech coding 解釋
語間編碼
  • 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。
  • coding : n. 編碼；譯成電碼。
  1. 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。
  2. 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。
  3. Thirdly, the paper discusses the driver of the peripheral equipment, how to port the uc / os - n and uclinux, h. 323 protocol and the application of the system in the digital speech classroom. also some software and hardware measure are adopted to enhance the system stability. at last, the shortcoming and the something to be improved are given. dsp can be used to realize real - time speech coding algorithm, and after porting ( ac / os - n, arm can manage the keyboard, the lcd and the ethernet peripheral etc. then the embedded network system with specific purpose can be used in others fields, such as pda, set of top, web tv, ect

    在實際設計實現中，為提高系統軟、硬體整體穩定性和可靠性，使用了以下幾種方法： （ 1 ）低電壓復位、抗電源抖動能力、增加時鐘監測電路、抗電磁干擾能力、散熱等技術； （ 2 ）多層pcb設計，線路板結構緊湊，電源部分採用數字5v 、 3 . 3v 、 3v 、 1 . 8v和模擬5v多電源供電； （ 3 ）選用表面貼和bga封裝的器件； （ 4 ）按照軟體工程的要求進行系統分析，規劃系統框圖、流程分析、模塊劃分，減小了不同模塊的相關性，從而最大限度避免了錯誤的發生。
  4. Second, we optimize the codebook and choice a part of the codeword which is used most efficiently. the result is not degraded too much while the complexity is reduced. at the end of the paper the development prospect of cs - acelp and speech coding are described

    對lsp參數量化中的第一級碼書的128個碼字的使用頻率進行了統計試驗，選用了128個碼字中使用頻率高的112個碼字作為新碼書，語音質量基本不變但降低了碼書搜索的復雜度。
  5. 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。
  6. Currently, research focuses of industry include speech coding with low bit rate, low delay, low complexity and good perceptual quality

    比特率低、時延小、復雜度低、音質又好的編碼方法是目前國內外民用工業研究的熱點。
  7. 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
  8. Speech communication is one of the most used modes in the digital trunking communication system. excellent algorithm of speech coding can save the bandwidth resource, improve the utilization of frequency, so it has important value for investigation

    語音通信是數字集群通信系統中最常用的通信方式之一，優良的語音編解碼演算法能夠更加有效地節省帶寬資源，提高頻率利用率，因此具有重要的研究價值。
  9. 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
  10. Speech coding systems are generally based on narrowband speech at present, but its frequency is restricted in 200hz ~ 3400hz and its sample rate is 8khz. with the development of the wideband speech, its bandwidth which is from 50hz to 7khz causes the quality of speech communication to approach in the feeling of face - to - face conversation, and makes the speech natural, expressive and comfortable. hence, it ’ s quite significant in researching on wideband speech coding system. in recent 20 years, the dsp and its software development kit has improved greatly, but the price has fallen sharply, thus it has more and more widespread applications now

    隨著當今世界的飛速發展，寬帶語音越來越受到人們的青睞，因為它的50hz 7khz的帶寬使得語音通訊質量接近於面對面交流的感覺，大大提高了語音的自然度、表現力和舒適度。因此，開發研製基於寬帶語音的編解碼系統具有十分重要的意義。在過去的短短二十年裡， dsp處理器的性能得到很大的改善，軟體和開發工具也得到相應的發展，價格卻大幅度地下降，從而得到越來越廣泛的應用。
  11. Finally, this paper discussed the software system design based on above hardware platform, including the dsp initialization program, the interrupt request service and speech coding

    最後討論了基於上述硬體系統的軟體設計，主要分為三個部分: dsp的初始化程序、中斷服務程序和語音編解碼程序。
  12. 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。
  13. The main aim of speech signal processing is speech coding, speech recognition and speech understanding by automaton. in this paper, firstly, some basic signal process are discussed, such as signal filter, sampling, fft, pitch detection, then, they are tested on the testing board designed by the author

    論文中首先對語音信號的基本處理問題進行了分析和對比，然後在自己設計的基於tms320vc5402的dsp實際系統上，進行了語音處理過程的濾波、采樣、傅立葉變換和譜包絡提取的演算法實現研究，討論了在演算法的dsp實現方法，分析了運行實驗結果。
  14. 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。
  15. The result of simulating examination makes this system obtain good rebuilding speech coding quality which comply with g. 729

    模擬試驗結果表明，此編碼系統獲得了較好的重建語音質量，能夠達到g . 729語音編碼標準所要求的語音質量。
  16. For different applications, the audio part of mpeg - 1 provides three layers ; we select the implementation of the first layer on the ti tms320 dsp considering the complexity of the algorithm and the quality of voice. in first, the development and categories of speech coding and the standard of the mpeg series has been described

    Mpeg - 1的音頻部分給出了三個層次以適應于不同的應用要求，綜合考慮演算法復雜度和話音質量要求，本課題選擇了mpeg - 1layer語音壓縮編碼方法並研究了其在titms320c6204dsp上的實現。
  17. 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。
  18. The application of adaptation technology in speech coding

    自適應技術在語音編碼中的應用
  19. The demand is the power forcing speech coding to progress. traditionally linear prediction ( lpc ) vocoders are very efficient, which can encode speech from 800 to 2400bps, but unfortunately, artifacts such as buzzes, thump, and tonal noise always exist in them

    經典的線性預測（ lpc ）聲碼器具有很高的編碼效率，可以極低的碼率（ 800 2400bps ）對語音信號進行編碼，不幸的是它的合成語音聽起來很不自然，常常夾雜著嗡嗡聲，重擊聲或者音調噪聲。
  20. 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。
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