speech data compression 中文意思是什麼

speech data compression 解釋
語音數據壓縮
  • 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。
  • data : n 1 資料，材料〈此詞系 datum 的復數。但 datum 罕用，一般即以 data 作為集合詞，在口語中往往用單數...
  • compression : n. 1. 壓縮；壓緊；濃縮，緊縮。2. 加壓；壓抑。3. （表現的）簡練。4. 應壓試驗。
  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. At the receiving end, a inverse process is performed. the system receives low rate data and the fpga reorganizes a frame of data which is decoded by the compression chip every 20 ms. the obtained pcm signal is performed d / a to restore the analog speech signal

    在收端進行相反的過程，接收低碼率數據，並由fpga重新組幀，送至主晶元解碼得到pcm信號，再作d / a變換，恢復出模擬語音，系統是全雙工的。
  3. Carry on emulation to melp standard, realize that the compression of the pronunciation file is solved and pressed. first this thesis sample to wav file, carry on the speech to analyze and draws with the parameter to the speech data of every frame. these parameter include pitch, bpvc, jitter, lpc, etc. then, these parameters will be quantized by msvq technology

    該系統首先對語音信號進行采樣；按幀對語音數據進行語音分析和參數提取，提取的參數包括基音周期（ pitch ） 、多帶清濁音判別、非周期抖動標志、線性預測參數（ lpc ）等語音生成模型參數；接著對這些參數進行了量化，量化採用了多級矢量量化技術；最後在解碼端對各個量化參數進行解碼，利用這些參數結合語音合成模型重構語音。
  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. 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。
  6. 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。
  7. 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。
  8. 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。
  9. 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。
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