質量縮語 的英文怎麼說

中文拼音 [zhíliángsuōyǔ]
質量縮語 英文
quality code
  • 質 : Ⅰ名詞1 （性質; 本質） nature; character; essence 2 （質量） quality 3 （物質） matter; substance;...
  • 量 : 量動1. （度量） measure 2. （估量） estimate; size up
  • 縮 : 縮構詞成分。
  • 語 : 語動詞[書面語] （告訴） tell; inform
  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. The speech processing module can be divided into two parts, the first part includes speech compression, voice activity detection and echo cancellation module, which improve speech quality ; the other part includes dtmf and cpt module, which generate and detect some necessary telephony signal in the communication. this thesis is organized as follows

    該語音處理模塊由兩部分組成，一方面是對語音的處理，包括語音壓縮模塊、靜音處理模塊和回聲消除模塊，主要為了提高voip的語音質量;另一方面是對電話通信的控制和處理，包括雙音多頻模塊和呼叫進程音模塊，主要為了產生和檢測ip電話通信中一些必須的電話信號。
  3. To improve the speech quality, it is necessary to adopt such speech processing techniques like speech compression, voice activity detection, echo cancellation, jitter buffer and so on

    為了提高語音質量，需要採取一系列的語音處理技術，主要包括語音壓縮技術、靜音處理技術、回聲消除技術、抖動緩沖技術等。
  4. 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。
  5. 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。
  6. The bandwidth is a principal factor that makes the speech quality bad. it is also the most difficult problem to be resolved for the internet

    帶寬是影響語音質量的一個主要因素，也是因特網當前最難解決的問題，一個解決的辦法就是採用高效的語音壓縮演算法。
  7. 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。
  8. 4. aimed at the reduction of bit rate and the improvement of speech quality, a serial of speech coding schemes are studied in a gradual refinement way, and an integrated coding scheme at 1. 5 - 2. 4kbps is presented finally

    圍繞編碼位率的降低和語音質量的提高，以逐步求精層層遞進的方式研究了一系列壓縮編碼方案，並最終提出一個位率在1 . 5 2 . 4kbps的綜合編碼方案。
  9. As an important field, which influences the quality of ip phone, the quality of speech codec becomes the focus of research. the g. 729 is an 8kbit / s speech - coding standard issued by the international telecommunications union ( itu )

    G . 729是國際電信聯盟（ itu ）頒布的編碼速率為8kbit s的低速率語音壓縮編碼標準，它採用了共軛結構算數碼本激勵線性預測（ cs - acelp ）技術，可以達到32kbit s的adpcm的語音質量。
  10. For users of digital cameras, file compression is an issue you must be aware of if you want to be able to view and share your images

    數碼相機使用者如果希望拍攝出高質量的圖片，就必須注意文件壓縮這個術語。
  11. System scheme of speech coding plus spread spectrum communication was presented based on a full analysis of noise characteristic, attenuation characteristic and impedance characteristic of low - voltage power line. spread spectrum carrier ( abbreviated as ssc ) technology is adopted to overcome problems existing in signal transmission over power line. high quality, low rate mbe compression algorithm was used to complete speech encoding and decoding

    在對低壓電力線路的噪聲特性、衰減特性和阻抗特性三個方面充分分析的基礎上，本文提出一種語音編碼+擴頻傳輸的系統總體方案，採用擴頻載波（ spreadspectrumcarrier ，縮寫為ssc ）技術克服電力線傳輸信號存在的問題，採用語音合成質量高並具有較低碼率的mbe壓縮演算法完成語音信號的編解碼。
  12. Now, the standards of speech compression coding provide a way of transporting speech signals efficiently. in fact, all of them are to reduce the baud rate of data under definite speech quality

    相繼出現的語音信號壓縮標準為語音信號的高效傳輸提供了一種有效方法，其實質就是在相當的語音質量指標下，降低數字化語音的數碼率。
  13. 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。
  14. 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。
  15. Vocoders based on the mbe model are popular due to their good synthesis speech quality

    基於多帶激勵模型的語音壓縮編碼演算法以其較好的合成語音質量而受到廣泛的重視。
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