語音數據壓縮 的英文怎麼說

中文拼音 [yǔyīnshǔjùyāsuō]
語音數據壓縮 英文
speech data compression
  • 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
  • 數 : 數副詞（屢次） frequently; repeatedly
  • 據 : 據Ⅰ動詞1 （占據） occupy; seize 2 （憑借; 依靠） rely on; depend on Ⅱ介詞（按照; 依據） according...
  • 壓 : 壓構詞成分。
  • 縮 : 縮構詞成分。
  • 語音 : speech sounds; pronunciation; voice
  • 數據 : data; record; information
  1. The basic characteristics of the current data network are point - to - point, connectless, doing one ' s endeavor, no quality of service, etc. these characteristics do not meet the requirement of real - time services, therefore, the realization of voip need support of the some key technology. these technologies includes : speech sound coding and data compression, real - time transmission and control, mute compression and multicast, acoustic - echo cancellation and comfort noise generator, dynamic monitor and guarantee of quality of network service, as well as, the compatible of different network and different protocol with each other

    但現有的數據網路的基本特性：點對點的、無連接的、盡力而為的、沒有服務質量保證等特性並不適合與實時的業務要求，因此voip的實現需要一些關鍵技術的支持，這些技術包括：語音編碼和壓縮技術、實時傳輸和控制技術、組播技術、靜音壓縮和舒適噪聲生成技術、回聲消除技術、網路服務質量的動態監測和保證技術、以及不同的網路、不同的協議之間的互連互通等等。
  2. It can provide many functions : the conversion between transfer formats, the conversion between communication protocols, the conversion between analog signals and digital signals, the conversion of control commands, real time audio code / decode, ip call control, control of rtp / rtcp, call detail logs, accounting and some of gatekeeper ' s functions such as address tansfer / search.

    它提供以下功能：傳輸格式間的轉換，通信規程間的轉換，音頻、數據格式轉換，呼叫控制信令的轉換，對pstn的呼叫控制功能， pstn電路交換功能，實時語音處理語音壓縮功能， ip網路呼叫控制功能，對實時對話協議rtp rtcp的控制功能，呼叫日誌，計費功能，還包括關守的地址翻譯搜索等功能。
  3. Of course, the theories and approaches about the design of database management system and how to integrate our previous research results into the fpax ' s system will be discussed as a main point. part two : in this part, we attempt to introduce the model of the web site so as to build a teleconsultation system based on the browser / server model. via using the java language program, we hope that our telemedicine user can achieve the goal of the teleconsultation just using a www browser

    本研究試圖對應用java技術構築遠程醫療網進行一點初淺的探討，希望藉助java的強大功能，應用jsp / servlet建立一個遠程醫療網站;應用javaapplet實現電子白板、語音傳輸、文字交流等功能;採用jpeg和小波演算法實現圖像的壓縮;實現多種傳輸模式結構;實現病歷數據的后臺管理;最終建立一個適合不同操作系統的遠程醫療會診及教育系統。
  4. Now we can transmit data, voice, and fax in communication lines whose bandwidths from 9. 6kb / s to 256kb / s through the voice / data multiplexing system. this method is good for improving bandwidth utilization and used widely in telecommunication, finance and military

    語音擻據復接設備可以實現在一條帶寬為9 . 6kb s 256kb s的用戶數字線路上實時傳輸數據、壓縮語音傳真的綜合業務，是利用帶寬的有效手段，在電力、電信、金融和軍事領域應用日益廣泛。
  5. This article describes the plc ’ s using enviroment and the users ’ needs in details, and then tells the differences between the dplc and the troditional plc. it also describes how to decimation and compress the tele - voice, how to deal with the data tunnels, and the multiplex data through the the line which ’ s bandwidth is changeable between 10. 8bps and 28. 8kbps this project uses the programmable vopp chips, the cpld for bus controlling, the fpga and the mcu with high processing ability. all of these ensure the high ability to process the signals and the operations

    本文詳細介紹了高壓電力線載波機的使用環境和用戶需求，全數字電力載波機和傳統模擬載波的的差異;按電力調度的需求設計完整的兩路語音、兩路數據的電力線載波通訊機。詳細描述電話語音的采樣，壓縮;數據通道的處理;及在10 . 8kbps 28 . 8kbps可變線路速率通道上的數據復接處理。從實現方法上看，該項目採用專用可編程vopp語音處理晶元， cpld晶元做為總線控制，大規模fpga集成電路，高處理能力的mcu 。
  6. This paper mainly discusses the design principles and chief techniques of a digital accessing system for power - line communication net ( plcn ). the technology of low bit rate speech compression high - speed modem based on plcn adaptive equalization to the channel anti - jamming and anti - fading are applied in this system. so speech tele - control data and tele - protection signals can be transmitted high quality in the band - limited channel

    該系統綜合應用了低比特率語音信號壓縮編碼技術、基於電力通信網的高速調制解調技術、信號傳輸的通道自適應均衡技術和抗干擾、抗衰減技術，可在帶限通道中高質量的傳輸語音、遠動數據和遠方保護等信號，具有較高的整體性能。
  7. The characteristic and key technologies of the system are as follows : ( 1 ) in realizing the live broadcast of audio and video, the problem of immense multimedia data and low networks bandwidth utilization ratio is solved by using mpeg - 4 as format of audio and video data. audio and video data are collected by video card cv500 which developed by beijing sum tone company ; meanwhile, the contradictory between the delay of networks transmitting and the quality of the image is well solved by setting a " bi - buffer area "

    系統實現中解決的關鍵問題和特色主要有以下幾個方面： （ 1 ）在視音頻直播功能的實現中，通過使用北京算通公司的cv500視頻採集卡和cv500sdk進行視音頻數據採集，並採用當今最新的圖像和語音編碼壓縮標準mpeg - 4作為視音頻數據的採集格式，既保證了圖像的質量，又大大縮減了視音頻所佔的帶寬，從而解決了多媒體數據量大、網路帶寬利用率低的問題；同時，通過設置環形緩沖區的辦法來調和網路傳輸延時與圖像質量之間的矛盾，取得了較好的效果。
  8. However, there are some questions about instant messenger : firstly, instant messengers existed ca n ' t support to draw instantly graphs for communicating with others because there is no environment to draw graphs in the local computer in instant messengers. secondly, or can do it, but for transferring too large picture encode data, the instant messengers are only running in local area networks, not in internet

    現有的即時通信系統在支持圖形即時雙向交互方面還存在著如下問題： （ 1 ）不能支持基於圖形的即時通信，在即時通信系統中缺少即時繪制圖形的環境，只支持基於文字、文件、語音和視頻的通信； （ 2 ）部分支持基於圖形的即時會話的系統，傳輸的數據是圖像壓縮數據，單位時間內傳輸的數據量大，不適合在開放網際網路上廣泛使用。
  9. Conjugate structure algebraic code excited linear prediction was approved as itu recommendation in 1996 based on the project of usa at & t, japan ntt, franc telecom and canada sherbrooke university. cs - acelp based on adapt linear prediction is one of the most sophisticate algorithms in the field of low bit rate speech coding

    共軛結構代數碼激勵線性預測語音編碼（ conjugatestructurealgebraiccodeexcitedlinearprediction簡稱cs - acelp ）演算法是1996年國際電信聯盟（ itu ）根據美國at & t 、日本ntt 、法國電信和加拿大sherbrooke大學聯合提出的方案而制定的，它是最復雜低數碼率語音壓縮演算法之一。
  10. This paper introduces a project of the wireless data transferring and the realization of speech encoding / decoding arithmetic based on the embedded system. in embedded system based on arm ? cpu, we accomplished the update of the system data by using the paging system, and emphatically researched how to avoid bit error. and, realizes the speech compression and decompression based on itu - t g. 729a, implement the speech synthesize of personal paging

    在以arm7為處理器內核的嵌入式系統上，通過尋呼系統實現了系統數據的無線動態更新，重點解決了尋呼誤碼造成的數據錯誤等問題；以itu ? tg . 729a語音編解碼標準為基礎，通過語音壓縮與解壓演算法實現了個人尋呼的語音合成。
  11. In order to deal with this problem, this paper introduces the author ' s research on some techniques related to speech processing, mainly including three aspects as follows : [ 1 ] in chinese pronunciation, each syllable contains the vowel, the vowel ' s length is the main part in the syllable but the vowel does n ' t contain the important information. according to these characteristics, we propose a method of adjusting the speech velocity by using similar waveform that is found by correlative coefficient in vowel part to lengthen or reduce the vowel part

    本文主要介紹了作者針對這一問題所作的關于語音調整的技術與方法的研究工作，其中包括（ 1 ）根據漢語語音發音時每一個音節都含有母音，母音長度占音節長度的主要部分但是卻不包含發音的主要信息這些特點，提出在語音的母音部分利用相關系數尋找相似波形，然後對母音部分進行幾個相似波形的壓縮或擴展的方法來改變母音的長度進而調整語速。
  12. Rapid development of data business, growing of packet network technology, and increasing of communication channel capacity, etc, bring this problem the answer : the next generation network will be base on the ip, and it will be to consist of network architecture which are diverse, synthetic and open such as speech sound, data, multimedia etc. the principle of voip ( voice over internet protocol ) is not complicated : at the sending end, sample the analogue speech sound signal, code and compress, then package and transmit it over the packet network

    數據業務的快速發展、分組網路技術的成熟、數據網路通信通道容量的不斷增加等給這個問題提供了答案：下一代網路將是基於ip的，下一代網路將是可以提供包括語音、數據和多媒體等各種業務的、綜合的、開放的網路構架，而voip正是這個答案的具體體現。實現voip的原理並不復雜：將模擬的語音信號采樣、編碼並進行壓縮，封裝在數據網路的分組中進行傳輸，在接收端對數據進行解碼、數模轉換恢復成模擬信號即可。
  13. And the implementation of lower voice data transmission is the core of the whole software, which involves voice input and output, voice sampling and playback, signal encoding and decoding, signal compressing and

    底層的語音數據傳輸實現，是整個軟體的核心，包括:語音輸入輸出、語音採集回放、語音編解碼，信號壓縮和解壓縮、信號加解密，數據發送和接收。
  14. Then, the paper discusses the realization of the speech ' s sampling module, playback module, communication module, compression and decompression module in detail, and gives a self - adaptive silence detection algorithm and a policy to make disorder data package in order

    文章詳細地討論了語音網路通訊的採集模塊、回放模塊、通信模塊和壓縮解壓模塊的具體實現，並提出了一種自適應的靜音檢測演算法和數據包亂序調整的策略。
  15. The paper gives a project to realize speech communication between teacher and student in network, which is based on speech ' s compression and decompression technology, network communication technology and multithreading technology. the point - to - point full duplex and real time speech communication in network is implemented on lan by using peer to peer model, udp communication technology and simple data - compression method, and programming to the full duplex sound card

    以語音壓縮編解碼技術、網路通信技術和多線程技術為主要理論基礎，給出了教員與學員間語音網路通訊的具體實現方案：採用對等模式及udp通信技術和簡捷的數據壓縮技術，對全雙工聲卡編程，在局域網內實現點到點的全雙工的、實時的語音通訊。
  16. In the description of circuit design, the emphasis is paid the following hardware modules : ad / da inverter, dsp module, external program / data memory, cpld control logic, serial communication module, power module, and so on. problems and the corresponding solutions found in the design and debug stage are discussed, too. finally, the low - level software driver design is presented in detail, including system booting, initialization of dsp registers, cpld logic and timing control, drivers for asynchronous communication fifo, and drivers for ad converter

    在電路模塊分析中，重點介紹了語音的輸入放大和輸出緩沖部分、 ad da轉換、 dsp語音壓縮解壓、外部程序數據存儲器、 cpld邏輯控制、串列收發組件、電源供電以及dsp的jtag介面等等，並且給出了在硬體電路設計和調試過程中的問題與解決辦法。
  17. In the realization of foundation, this paper discusses windows multithread processing and realization process of socket thoroughly, and researches working flow of voice network transmission. then the paper unifies preliminary audio function and callback function, and designs the data structures which can be used in voice transmission, in this way, audio data collection and audio broadcast can be realized. finally, the paper gives realization steps of audio compression and decompression

    2 .在底層實現中，深入探討windows多線程處理和套接字實現過程，研究語音網路傳輸的工作流程;結合低級音頻函數和回調函數，設計音頻傳輸中將用到的數據結構，實現音頻數據採集和音頻播放;具體給出音頻壓縮和解壓縮的實現步驟。
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