speech channel 中文意思是什麼

speech channel 解釋
電話通路
  • 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。
  • channel : n 1 水路，水道，渠，溝；海峽；河床，河底。2 （柱等的）槽，凹縫；【機械工程】槽鐵，凹形鐵。3 〈比...
  1. Digital speech technology has been widely used in many fields of communications in recent years, but it is just at beginning to use acoustical wave to implement underwater digital speech communications, which mainly results from the bandwidth - limited underwater acoustic communications channel and its high temporal and spatial variability

    目前，數字語音技術已廣泛應用於許多通信領域，但是在水下利用聲波進行數字語音通信卻是剛剛起步的新技術，這主要是由於水聲通道有限的通信帶寬及其時變、空變等特性所決定的。
  2. European digital cellular telecommunications system - half rate speech - part 4 : comfort noise aspects for the half rate speech traffic channel ; english version ets 300581 - 4 : 1995

    歐洲數字移動電話遠程通信系統.半價話務.第4部分:半
  3. It link up to good at, is it make pieces of expansive speech house to mean while being what is called, the remarkable leader is clear, it must be a reciprocal two - way channel to communicate, could get the result of communicating

    所謂善於溝通，並不是指做個滔滔不絕的演說家，卓越的領導者明白：溝通必須是條有來有往的雙向管道，才能達到溝通的效果。
  4. We shall see how a speech channel be conveyed as a series of amplitude values, each value being represented as a sequence of 8 binary digits

    被省略的邏輯主語多數情況下指的是主句的主語或逗號前整個句子的含義，點也有少數列外。具體情況以後結合課文講解。
  5. 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。
  6. First, the traditional speech detection method based on short - time energy is discussed, including its principle and implementation. then it is used for the jumping - off point detection of speech signals transmitted by awgn channel. simulation results are provided

    包括該方法的原理、實現，並將其應用於加性高斯白噪聲通道干擾下的語音信號起點檢測實驗，給出了實驗的統計結果。
  7. In his speech, jiang jufeng said the festival is an important platform for displaying the newest technologies and products in the field, an ideal channel for program and equipent trade and exhibition, and a window of sichuan for foreign exchanges and cooperation

    蔣巨峰在致辭中說，四川電視節集中展示了當今世界電視業發展的新成就，為電視界搭建一個良好的合作平臺，為影視節目和設備交流提供一個重要的互通渠道，更為四川開啟了一個對外宣傳和學習交流的窗口。
  8. By reducing coding rate, more speech signals can be transferred in the same channel. so, low bit rate speech coding has especially important significance when the transmission rate is limited very strictly

    通過降低編碼速率，可以使同樣的通道容量能夠傳輸更多路的語音信號，在傳輸比特限制十分嚴格的場合，低速率語音編碼具有特別重要的意義。
  9. In this paper, we give a detail discussion on the key technology, including software and hardware designing of g. 729a multi - channel speech codec realtime implemention on a simple dsp processor - tms320c6202. in combination with the requirements of a military communication network, the atm adaptation solution of g. 729 coder bit stream is analyzed. a kind of new atm adaptation technology - aal2 is introduced. the analyse and research of aal2 are provided

    本文詳細討論了多路g . 729a語音編解碼器在一片dsp處理器tms320c6202上實時實現的軟硬體設計和關鍵技術。結合某軍事通信網設備的需要，進而對g . 729語音編碼的碼流的atm適配方案進行了分析。提出了用一種新的atm適配技術- - aal2進行適配的方案。
  10. Therefore, in this degree paper, designing and applying an equipment to detect and record the locomotive operator ' s speech information is mainly concerned. the paper is made on the basis of two projects the author involved, namely, mvr01a multi - channel digital speech recording system and meishan locomotive speech recording and querying system

    本學位論文的實踐基礎建立在作者所參與的兩個項目： 「 mvr01a多通道數字語音記錄系統」和「梅山機車語音記錄查詢系統」之上；論文將主要著重於介紹如下兩個方面內容：車載語音記錄系統的設計與實現。
  11. Since an underwater acoustic communication channel is bandwidth - limited, transmission of quantized speech samples at high bit rates is restricted, hence speech signals must be compressed

    要想在水中進行數字語音通信就必須對語音信息進行大幅度壓縮，降低傳輸所需的比特率。
  12. Single channel subtractive - type algorithms is often used in the speech enhancement, but musical residual noise will appear

    處理寬帶噪聲最通用的技術是減譜法，但在聽覺上形成殘留「音樂噪聲」 。
  13. Another new speech detection method using the ad aline neural network is discussed, including its structure, learning algorithm and simulation results for the detection of speech signals transmitted under different channel conditions

    討論了採用自適應線性神經元網路adaline的方法檢測語音起點。包括該網路的結構、學習演算法以及對不同干擾條件下語音信號起點檢測的統計結果。
  14. The important parts are the applied hardware design and software about it. the manuscript drawings, software glow chart about speech input / output and channel interface are offerd. these part questions faced and settle methods about debugging are expatiated too

    具體給出了設計的原稿圖紙、語音輸出受入程序設計流程，通道介面輸入輸出程序設計流程，並在最後闡述了實際調試過程中遇到的部分問題和解決問題的方法。
  15. Directly digitizing of speech signals needs too much high bit rate in digital speech communication, thus compression and coding of the speech signals is essential in the process to increase the efficiency of transmission and storage, as well as the channel capacity

    數字語音通信中，語音信號直接數字化所需的數碼率太高，為了提高傳輸和存儲的效率，充分利用通道容量，必須對數字語音信號進行壓縮編碼。
  16. At the beginning of this thesis, we introduce the fundamental of the acoustics and the perceptual mechanism. next, different kinds of speech processing methods including time processing and time - frequency analysis are presented, such as short time average energy, short time cross zero analyses, short time autocorrelation function analyses and fft. at last, we focus on the sound separation, especially on single channel sound separation

    在這篇文章開始的部分，我們介紹了聲學的基礎知識和人類聲音感知的機理;接下來，我們給出了在時域處理和頻域處理語音信號的一些經典的技術，比如短時平均能量分析、短時過零分析、短時自相關函數分析、快速傅立葉變換等;本文重點從理論和實驗上討論語音分離，特別是單聲道語音分離的演算法及其在分離音樂鼓點中應用。
  17. 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。
  18. The accuracy of a speech recognition system in actually noisy environment is seriously affected by the additive noise and the channel distortions

    實際環境中的背景噪聲和傳輸通道變化所引起的畸變嚴重影響了語音識別系統的性能。
  19. This system uses the evm of dual - core tms320vc5471. for the different core configuration, the dsp has a / d and d / a modules to implement speech signal i / o channel, real time implementation of speech coder. meanwhile embedded operation system uclinux is transplanted into it

    結合cpu兩個內核不同的體系結構，在dsp內核上連接a / d 、 d / a模塊實現語音信號i / o通道以及實時實現語音編解碼演算法，在arm內核上移植了clinux嵌入式操作系統。
  20. The speech quality produced by g. 729a is equivalent to that of 32 kb / s adpcm for most operating conditions, achieving the request of long - distance telephone. these conditions include channel errors, multiple encodings

    729a具有非常好的性能，其語音質量與32kb s自適應差分脈沖編碼調制（ adpcm ）演算法相當，達到了長途電話質量要求，在有隨機比特誤碼、發生幀丟失和多次轉接等情況下有很好的穩健性。
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