語音編程系統 的英文怎麼說

中文拼音 [yǔyīnbiānchéngxìtǒng]
語音編程系統 英文
voice programming system
  • 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...
  • 程 : 名詞1 （規章; 法式） rule; regulation 2 （進度; 程序） order; procedure 3 （路途; 一段路） journe...
  • 系 : 系動詞（打結; 扣） tie; fasten; do up; button up
  • 統 : Ⅰ名詞1 （事物間連續的關系） interconnected system 2 （衣服等的筒狀部分） any tube shaped part of ...
  • 語音 : speech sounds; pronunciation; voice
  • 編程 : c programming
  • 系統 : 1. （按一定關系組成的同類事物） system 2. （有條理的;有系統的） systematic
  1. This thesis tries to update the cmdsr system to achieve the characters below : real - time, better robust, higher recognition rate, non - special - man. considering the disadvantages of traditional improved spectrum subtraction speech enhancement, this thesis proposes the theory of fuzzy spectrum subtraction based on the fuzzy theory and improved spectrum subtraction speech enhancement ; as for the difficulties of detecting the endpoint of speech signal, the thesis gives the table of initial and the improved parameters, with which we can confirm the endpoints of mandarin digit speech ; the thesis puts forward two - level digit real - time speech recognition system, the first level is based on discrete hidden markov model which is linear predictive coding cepstrum ( lpcc ) and difference linear predictive coding cepstrum ( dlpcc ), the second level is based on formant parameters ; as for the realization of hardware, the thesis depicts the realization of every part of cmdsr based on the tms320vc5402 in detail ; as for the development of software, the thesis gives the software design flow chart of cmdsr, simulates the basic theory with matlab language and gives the simulation results

    針對傳統的「改進譜相減法語音增強」參數設定單一、環境適應能力差的缺點，提出了一種利用模糊理論和「改進的譜相減法」結合的「模糊譜相減法語音增強」 ;針對語音信號端點檢測困難的特點，通過matlab模擬試驗，給出了能夠準確確定數碼語音端點的初始和改進參數表;提出了利用基於線性預測編碼倒譜參數和差分線性預測編碼倒譜參數相結合的離散隱含馬爾可夫模型進行第一級識別、利用共振峰參數進行第二級識別的兩級漢語數碼語音識別系統，在保證系統實時性的同時，實現連接漢語數碼語音識別系統識別率的提高;在硬體實現上，詳細闡述了基於tms320vc5402的連接漢語數碼語音識別系統各部分硬體設計;在軟體開發上，給出了連接漢語數碼語音識別的軟體設計各部分的流程圖，並對各部分進行了matlab模擬，並給出了模擬結果。
  2. The query module is a retrieval system on internet, it adopts b / s mode of three - layer structure, and has been implemented by using the inner component ado and the third party component fileup of asp technology and sql server 7. 0. this module can support the transmission of multi - media file. the discussion module is a electronic white board which is based on network, it has been implemented by java language and java media framework ( one of java media apis )

    本答疑係統由查詢和討論兩部分組成，查詢部分是一internet上的全文檢索系統，它採用三層結構的b / s模式，利用asp技術的內置組件ado 、第三方組件fileup和sqlserver7 . 0來實現，它能支持與問題相關的多媒體文件的上傳和下載，從而為查詢答疑提供了與問題相關場景（即音視頻信息）的支持；討論部分則是一網上實時交互系統，該系統是以java語言和javamediaapis （ applicationprograminterfaces應用程序介面）提供的jmf （ java多媒體框架）編程實現的網路電子白板，通過它能在ip網路上實現文字、圖形、圖象、音頻視頻信息的實時交流，使網上答疑變得直觀生動和高效
  3. Once has the bandits and thieves to intrude guards against the place, the detector launches the wireless coded signal immediately, the networking center number which installs when is apart from defense area 150 meter within the main engine to send out the police whistle sound to report to the police immediately, reports to the police dials to establish in advance or reports to the police the telephone, the handset number, answers in the police telephone to return puts user pre - record to report to the police the pronunciation, long - distance reports to the police, simultaneously comes the real - time transmission through the internet to deploy troops for defense, to withdraw from a defended position, to report to the police and so on the condition, inquires the historic record through the computer network

    該系統還採用美國進口原裝晶元與先進的無線數字高頻技術微電腦cpu控制器主機組成。在防範地點安裝好主機后，並設置在布防狀態。一旦有盜賊闖入防範地點，探測器立刻發射無線編碼信號，安裝在距防區150米以內的主機立即發出警笛聲報警，報警時撥打預先設定的聯網中心號碼或報警電話手機號碼，接警電話里回放用戶預錄的報警語音，遠程報警，同時通過網際網路來實時傳遞布防撤防報警等狀態，通過電腦網路來查詢歷史記錄。
  4. 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。
  5. 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。
  6. Liss information technology - programming languages, their environments and system software interfaces - guidelines for the preparation of language - independent service specifications

    信息技術.編程語音語言環境和系統軟體界面.語音獨立服務規范制備指南
  7. The card control adopts the methods as multiple thread programming, socket communication, queueing, and realizes call routing, fixed _ time or real _ time dialing out. multilateral communication, and voice response, so that satisfys telephone system " s real _ time request. in the end, the thesis prospects the future call centers, such as internet call center, multimedia call center, their framework and key technology

    在對板卡控制上採用了多線程編程技術， socket通訊技術，排隊技術相結合的軟體系統方案，實現了呼入呼出的智能的呼叫路由、即時定時外撥、多方通話、語音響應同步錄音等功能，滿足了電話系統的實時性要求。
  8. These systems operate on a wide variety of continuous time signals, which include speech, medical imaging, sonar, radar, electronic warfare, instrumentation, consumer electronics and telecommunications systems ( terrestrial and satellite ). one of the key to the success of these systems has been advances in the development of the font end of the electronic systems - analog - to - digital converters ( a / d ) which converter the continuous - time stimuli signals to discrete - time binary - coded form

    這些系統可廣泛地應用於處理連續時間信號，包括語音、醫學成像、聲、雷達、電子對戰、儀器、消費電器、遠程通訊（地面和衛星）等，而這些系統成功的關鍵因素之一就是電子系統的前端部件? ? a d取得了長足的進步（ a d把連續時間信號轉換成離散時間、二進制編碼的數字信號，便於后級精確的數字信號處理） 。
  9. In the next, we discuss the system of the meg - 1 layer i. the paper centers on the two kernel sub - parts : filtering coding and psychoacoustic model, do some research work in sub - band coding ( cbc ) theory and the relate theory such as quadrature mirror filter ( qmf ) and analyse sub - band filter ; also do research work in psychoacoustic theory especially the part related to the mpeg - 1 layer i. in the third chapter, introduce the ti tms320c6000 series dsps and their characteristics, also about the software development flow and the ti dsp / bios operating system of it. the forth chapter is the most important, firstly, according the algorithm flow in protocol, using c language validate the algorithm ; then, transplant and optimize the coding in dsp. in the processing of optimize, acording the assembler program characteristic of ti dsp, the paper put forward the analyse sub - band filter dsp optimization algorithm base on the eight spot idct. the algorithm has been optimize have greatly improved the work efficiency. make use of the technology of the dsp / bios host channels, data io pipe, software interrupt, we implement the musicam algorithm base on dsp / bios

    論文首先對當前語音編碼技術的發展、分類以及mpeg系列音頻標準作了介紹；接著在第二章，給出了layer的musicam （ masking - patternuniversalsubbandintegratedcodingandmultiplexing ）演算法的系統組成，圍繞分析子帶濾波器和心理聲學模型兩個核心模塊，深入研究了子帶編碼工作原理、比特分配及子帶編碼中用到的正交鏡像濾波器和分析子帶濾波器；探討了心理聲學基本原理和mpeg . 1layer所用到的心理聲學模型。第三章對titms320c6000系列dsp作了簡介，介紹了6000系列dsp結構特點、 c6000dsp軟體開發流程和tidsp / bios操作系統。第四章是本文的重點，首先根據協議給出的演算法用標準c語言編程實現並調試通過。
  10. The paper focuses on the following two aspects : the design and application of locomotive speech recording system researches, feelings, and views of the author about the project, mainly including : researches on speech coding ; views on the relative software engineering concept occurring in the " developing process of the project, such as how to guarantee the controllability of the developing process, the reliability of the lifestyle and issues concerning the reproduce of the project of same type. the paper traces a train of thought as follows : creating a system based on the requirement of the design, summarizing and analyzing the main necessary techniques and ideas to apply it, and the author ' s own understanding during the practice

    介紹作者在這個項目中的一些研究、感受與體會，集中在兩點：對語音編碼的研究；對開發過程中所牽涉到的軟體工程思想的實際體會（怎樣保證軟體項目的開發進程可控性與生命周期的可保證性、由項目實際引發而來的可傳承性問題）論文的總體寫作思路是：依據設計要求實現一個系統，總結、提煉並且分析系統實現中的若干種主要技術、思路和自己的實際開發感悟。
  11. After making a full thought on the previous computer phonetic processing and recognition technology, the author devotes the paper to the designation of phonetic samples, phonetic boundary detection, phonetic auto - segmentation, the design of phonetic distinctive feature database, the memory ways for store phonetic data, the contrastive contents about words and phrases on psc examine papers and the design of the system window. the author also offers a typical flow diagram and a program about using the visual studio software to draw procedures which plays an important role on the realization about the final success of the software system discussed on this paper

    在借鑒和參考目前計算機語音處理和識別技術的基礎上，著重對語音樣本的選取、語音端點檢測、語音自動分段、以及語音特徵數據庫的設計、語音數據的存儲方式、 psc試卷單字、詞語的對比內容、系統的界面設計進行了深入的探討和研究，詳細列出了一些有代表性的流程圖，提出了利用visualstudio可視化計算機軟體進行編程的具體方案，對本軟體的最終程序實現具有指導作用。
  12. The operating system and control program run on host processor mpc852t, the voice compress codec program run on the voice processor ac48304, they communication with the host port interface

    主處理器mpc852t上運行操作系統及控製程序，語音壓縮分組處理器ac48304進行語音壓縮編解碼。
  13. We adopt a ping - pong buffer mechanism to guarantee the system ' s real - time implementation. in the hardware design, we use adsp2188n and codec chip msm7702 to accomplish the algorithm and flash memory sst391f080 to store the startup code. assembly language code and some necessary initialization data

    在硬體設計中，本系統以adspzi88n為核心，結合codec晶元msm7702完成編解碼演算法，使用flash晶元sst39lto80來存儲系統的啟動程序、匯編語言程序和初始化表格數據，使用話筒和聽筒來完成語音的輸入輸出。
  14. In this dissertation, an embedded video monitoring system based on network is studied deeply, and then implemented the hardware device drivers to the chip of the vweb company ’ s vw2010. the design is based on the mpeg - 4 technology and embedded linux. the first three chapters of the thesis are to study the video surveillance system ’ s current background, main hardware structure and the main functions of software molds

    本論文的重點:研究了網路視頻監控系統的基本硬體體系結構和軟體功能模塊，提出了一種使用晶元vw2010來實現視音頻硬編解碼的驅動程序設計方法，該設計基於當前最流行的mpeg - 4編碼技術和開源的嵌入式linux操作系統;接著介紹了基於晶元vw2010的能兼容多語言的osd界面設計的幾個關鍵技術;論文最後給出了嵌入式linux下控制多種雲臺鏡頭的研究結果和設計方案。
  15. Secondly, the design of the wideband speech processing system based on dsp is described. finally, the haredware and software of the system is expatiated, which includes the design of system architecture and its intefaces ; the programming and debugging of software in dsp, the driver of the speech gathering card and the interrupt service of tms320vc5416

    接著講述了基於dsp的寬帶語音處理系統的總體設計，包括系統架構設計、演算法與晶元選型等。然後是寬帶語音處理系統的硬體組成與介面設計。最後是寬帶語音處理系統的軟體實現，包括dsp的數據接收與發送的實現， dsp主程序與採集卡驅動程序的設計、編程與調試， dsp中斷處理的實現等。
  16. In the end, original speech wave and rebuilding speech wave, original speech frequence and rebuilding speech frequence is listed

    程序最後顯示表明，此編碼系統具有較高的語音編碼質量，獲得了很好的重建語音質量。
  17. During the process of voice terminal design, the author firstly presents its hardware structure, including the coding and encoding module of ip, the interface to former access network equipment and etc, and then the author illustrates in detail the software realization of g723. 1 & g711 coder and encoder, then introduces how to realize the application of rtp / rtcp protocol in h. 323 gateway by openh323, including the design of the process and the realization of the data package format and partial code

    在語音終端設計中，首先提出了其硬體構架，包括ip語音編解碼模塊，與原接入網設備的介面等；然後重點討論了g . 723 . 1和g . 711編解碼器的軟體實現。在h . 323網關模塊中，首先討論了ip電話網關的系統架構；接下來介紹了openh323軟體開發包，並詳細介紹了利用openh323實現rtp / rtcp協議在h . 323網關中的應用，包括流程設計，數據包格式和部分代碼實現。
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