音頻術語 的英文怎麼說

中文拼音 [yīnbīnshùyǔ]
音頻術語 英文
audio terminology
  • 音 : 名詞1. （聲音） sound 2. （消息） news; tidings 3. [物理學] （音質） tone 4. （姓氏） a surname
  • 頻 : Ⅰ形容詞（次數多） frequent Ⅱ副詞（屢次） frequently; repeatedly Ⅲ名詞1 [物理學] （物體每秒鐘振動...
  • 術 : 術名詞1. （技藝; 技術; 學術） art; skill; technique 2. （方法; 策略） method; tactics 3. （姓氏） a surname
  • 語 : 語動詞[書面語] （告訴） tell; inform
  • 音頻 : [物理學] [電學] audio frequency; vf （voice frequency）音頻電路 voice frequency circuit; 音頻振蕩...
  • 術語 : term; onomastion; onym; terminology; technology; buzz word; nomenclature; jargon; technical terms
  1. Furthermore, the network technique only support the voice service cannot meet the communication requirements of people. people hope to obtain multimedia service such as packet data, video and picture phone etc besides voice service. all these need operators and researchers to look after the mobile communication scheme which optimizes spectrum efficiency and expands system capacity to accommodate more users in major metropolitan markets

    此外，僅支持語音業務的網路技術已不能滿足人們對信息交流的需求，人們希望能隨時隨地獲取除語音之外的數據、視頻和圖像等多媒體業務，這些因素都促使運營商和研究者尋求頻譜利用率更高、通信容量更大的移動通信解決方案。
  2. This paper describes a chinese speech audiometric system for clinical testing

    摘要本文描述了一種基於計算機音頻技術的漢語言語聽力測試系統。
  3. So acoustic echo cancellation ( aec ) technique has become a hotspot of competition in famous communicaton company all over the world. however, it is not always easy to implement acoustic equipments for communication system with satisfactory speech quality

    回波消除技術能有效地解決長距離電話網路、 ip電話、免提電話和視頻會議等通信系統中的回波問題，很好地改善了語音通信質量，具有廣闊的市場前景。
  4. Bluetooth operates in the global available 2. 4ghz free ism band, its topology is the piconet. it implements the wireless communication with data and voice, by the combination of circuit switching and packet switching

    藍牙使用免申請2 . 4ghzism頻段，採用微微網作為其網路的基本單元，並採用電路交換技術和分組交換技術組合，實現語音和數據的無線通信。
  5. 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
  6. This dissertation has discussed the r & d process of a broadcasting system based on network for fire fighting. this broadcasting system is based on the lan, and it has used the technology of voip. it has implemented the network speech communication by packet switching, and it has removed the shortcoming of the traditional broadcasting system which is based on circuit

    本文系統的論述了網路消防廣播系統的研製過程，該系統基於tcp / ip局域網路，利用了voip的相關技術，採用分組交換的原理來實現語音的網路傳輸，易於擴展，不受距離的限制，消除了傳統基於音頻電路消防廣播的弊端。
  7. Damping factor is a term audio enthusiasts ( and their rabid cousins, the audiophiles ) run into time and time again

    阻尼因子是一個音頻愛好者和發燒友反復追捧的術語。
  8. [ b ] damping factor [ / b ] is a term audio enthusiasts ( and their rabid cousins, the audiophiles ) run into time and time again

    阻尼因子是一個音頻愛好者和發燒友反復追捧的術語。
  9. Speech recognition and natural language process technology can receive and sort audio content. this is the first step toward retrieval entire video and sort video content. in this paper, we deal with the problem of segmenting news video data into semantically coherent scene using audio and video data, besides, classifying of news video content

    利用語音識別技術和自然語言處理技術對音頻流中的語音段進行處理，就可以解決音頻內容的提取和分類問題，這樣就更有利於檢索的進行，進而可以對所對應的視頻段進行內容分類，這些都為我們的研究創造了條件。
  10. 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。
  11. At first the article puts emphasis on analyzing those current network ip technology, various audio codec algorithms, realtime stream medium transmit technology and those process mechanism of realtime low bandwidth audio stream medium, etc. in allusion to high requirement of system, resulted from so many terminations of attending a lecture, rate of flow, bad situation of network and realtime interactive voice, a new algorithm and the relevant project of processing low bit - rate audio stream was brought forward

    本文著重分析了當前網路方面的ip技術，各種音頻編碼演算法，實時流媒體傳輸技術，實時低帶寬音頻流媒體的處理機制等等。針對網路實時應用中諸如客戶端眾多，各種多媒體數據流量大，網路狀況差，語音交互實時性等方面較高的要求，提出一種新的實時低帶寬音頻流處理演算法及相應的處理方案。
  12. First of all, the paper discusses protocols, standards, key technologies and work theories that relate with ip phone. it probes into the speech encoding technologies, sending and reception of data packet, recording and playing of audio frequency

    本論文首先討論了與ipphone相關的協議、標準、關鍵技術及其工作原理，然後探討了語音編碼技術，數據包的發送與接收，以及音頻錄制和播放。
  13. We have to keep the balance of these two factors. research of voip include : voice compression, system delaying, voice activation and voice priority. voip has to face another problem : system securities

    目前，網路語音通信技術的研究內容主要包括音頻壓縮編碼技術、系統延遲問題研究、靜音抑止技術、抖動補償技術和話音優先級技術等方面。
  14. The designed system includes functions of audio and video broadcast, screen image broadcast, interaction of electronic whiteboard, real - time courseware making, etc. the realization of the system relies on technologies of mpeg - 4, streaming media, network, database, smil and takes visual c + + of microsoft as the development tool

    所設計的系統具有視音頻廣播、屏幕圖像廣播、課件製作等功能。系統的實現綜合採用了mpeg - 4技術、流媒體技術、網路技術、數據庫技術和同步多媒體集成語言smil等，採用微軟的visualc + +作為開發工具。
  15. At first, the thesis briefly introduces the ip phone development status and standard, discuss its advantages and disadvantages, and then address the key technique issues of some commonly used speech compression standards

    本文首先介紹了ip電話的發展狀況和實現原理，分析了ip電話的特點和不足。接著論述了語音壓縮的關鍵技術，並對目前常用的一些音頻編解碼演算法作了簡要的介紹和比較。
  16. The system realization mainly includes the realization of recording, playing, coding - decoding, compressing and rtp of the audio as well as video capturing, replaying, etc. those problems of audio - compressing technology, audio - delaying, audio - jitter, echoing and voice activity detection which influence the audio quality have also been discussed and dealt with

    其中系統實現主要包括語音的錄制、播放、編解碼、壓縮、實時傳輸協議等的實現以及視頻採集、回放等。對影響語音質量的語音壓縮技術、語音時延、語音抖動性、回聲以及靜音等問題進行了探討和處理。
  17. I has mainly discussed the sofeware and hardware of aribrary waveform generator controlled by the microcontroller in my with the control of the high proformance microcontroller - 80c196kc, this instrument builds up the main circuit adopting single tone and dac of ad9857. in my project, i. have designed some special circuits with the advanced method of eda ( electron design automation ) based on vhdl and cpld. it can produce standard waveforms such as sine, square, triangle, and generate aribrary waveform according to the user ' s demand, at the same time, it has the ability to produce the amplitude modulation signal based on the carrier of sine and the aribrary waveform

    論文中主要對微機控制的任意波形發生器的軟硬體設計進行了相應的研究，該任意波形發生器以intel公司16位高性能單片機80c196kc作為控制器，分別利用了ad9857的單音頻模式和dac模式組成正弦波和任意波產生的電路，在硬體電路的設計中採用了先進的eda （電子設計自動化）技術，使用vhdl語言和可編程門陣列器件對一些特殊的電路進行了設計。
  18. 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。
  19. How to get the structure information and content meaning, make audio frequency signal having the same semantic classes is the key of the research of audio retrieval based on content

    如何提取音頻中的結構化信息和內容語義，使得無序的音頻數據變得有序，是基於內容的音頻檢索技術能否得以實用的關鍵所在。
  20. The audio information is sampled and measured and coded by the dsp card. the process speed is quick. there are a filter circuit, a power check circuit and a range - frequency characteristic check circuit which are designed to check the system status

    在硬體上，採用dsp語音處理技術，由高品質的dsp2000音頻卡進行音頻信息的採集、量化和編碼操作，不但提高了音頻的質量，也提高了音頻處理的速度。
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