speech network 中文意思是什麼

speech network 解釋
語音網路
  • 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。
  • network : n. 1. 網眼織物。2. (鐵路、河道等的)網狀系統,網狀組織,廣播網,電視網,廣播[電視]聯播公司。3. 【無線電】網路,電路。4. 【計算機】電腦網路,網。
  1. 21 callan e, kent d, guenther h, vorperian k. an auditory - feedback - based neural network model of speech production that is robust to developmental changes in the size and shape of the articulatory system

    但是要證實本文對人的語音生成和感知系統的闡述還需要許多定量的實驗。本文作者希望我們的研究能起到一個拋磚引玉的作用,引起更多的研究者的關注和興趣。
  2. Firstly, we study the construction of emotion - speech template database, and analyze the common features such as pitch, energy and formant. after choosing the useful features by using fuzzy entropy effectiveness analysis, we get better performance with the application of neural network. in addition, we propose some more efficient features such as speech rate, pitch slope, mel - frequency cepstral coefficients and its transient parameters, and design a processing model based on vector quantization for cepstral features to fusing different features

    本文首先介紹了情感語音數據庫的建立情況,然後研究了基音頻率、振幅能量和共振峰等目前常用的情感特徵在語音情感識別中的作用,並且通過一種基於模糊熵的特徵有效性分析方法進行了有效特徵的篩選,應用人工神經網路建立了初步的語音情感識別模型,經過實驗發現特徵篩選后系統的識別效果有著一定程度的提高。
  3. The president ' s speech was broadcast on a national television network

    總統演說由全國電視網聯播。
  4. The paper, which is based on the " research and development of the speed - up train ' s manipulation of optimization and train simulator " project, concerns the sound system in the train simulator. in the paper, the sound system includes two parts contents : firstly, improving the sound effect of the sound simulation system in the train simulator and enhancing 3d sound effect, such as sense of distance, sense of orientation, doppler effect etc, to make the sound of training environment more verisimilar ; secondly, realization of the speech communication in network between teacher and student in the train simulator

    本文是以鐵道部科技發展項目「提速列車優化操縱與機車模擬器研究開發」為依託,對機車模擬器的聲音系統進行研究:第一,對機車模擬器中聲音模擬系統的音效進行改進,增加3d音效,如聲音的距離感、方位感、多普勒效應等,使得訓練環境音響更加逼真;第二,實現機車模擬器中教員與學員間的語音網路通訊。
  5. Thirdly, the paper discusses the driver of the peripheral equipment, how to port the uc / os - n and uclinux, h. 323 protocol and the application of the system in the digital speech classroom. also some software and hardware measure are adopted to enhance the system stability. at last, the shortcoming and the something to be improved are given. dsp can be used to realize real - time speech coding algorithm, and after porting ( ac / os - n, arm can manage the keyboard, the lcd and the ethernet peripheral etc. then the embedded network system with specific purpose can be used in others fields, such as pda, set of top, web tv, ect

    在實際設計實現中,為提高系統軟、硬體整體穩定性和可靠性,使用了以下幾種方法: ( 1 )低電壓復位、抗電源抖動能力、增加時鐘監測電路、抗電磁干擾能力、散熱等技術; ( 2 )多層pcb設計,線路板結構緊湊,電源部分採用數字5v 、 3 . 3v 、 3v 、 1 . 8v和模擬5v多電源供電; ( 3 )選用表面貼和bga封裝的器件; ( 4 )按照軟體工程的要求進行系統分析,規劃系統框圖、流程分析、模塊劃分,減小了不同模塊的相關性,從而最大限度避免了錯誤的發生。
  6. 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。
  7. 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。
  8. Imagine a cell phone, for example, that can reconfigure its transceiver to use any network in the world and that at the push of a button can reprogram its processor to translate speech from one language to another

    舉例來說,想像這種行動電話吧,它能重新配置它的無線電收發器,來使用世界上各種網路,而且只要按一個鈕,就能重新設定它的微處理器,把一種語言翻譯成另外一種。
  9. At first system accomplishes chinese language automatic word segmentation and part - of - speech tagging through chinese input approach with word segmentation, then forms corresponding surface semantic network according to the semantic structure grammar, and finally gets corresponding data flow diagram and data dictionary according to the automatic generation algorithms of data flow diagram and data dictionary, the whole completion of the work, can not only provide a description environment of natural language for case, but also develop into the system which takes the question described on the basis of the natural language as the system ' s input

    工作的中心是自然語言篇章理解。系統首先通過分詞輸入法實現漢語自動分詞與詞性標注,然後根據語義結構文法產生相應的表層語義網路,最後根據數據流圖、數據字典自動生成演算法轉換為相應的數據流圖和數據字典。這項工作的徹底完成,不僅可以給case提供一個自然語言的描述環境,而且可進一步發展為基於自然語言描述問題作為輸入的系統。
  10. The primary advances in speech and audio signal processing that contributed to multimedia applications are in the areas of speech and audio signal compression, speech synthesis, acoustic processing, echo control and network echo cancellation

    語音和音頻信號處理的改進對多媒體應用的貢獻在下述范圍:語音和音頻信號壓縮、語音合成、聲學處理、回聲控制以及網路回聲消除。
  11. In the thesis the author presents some research on speaker recognition, mixed speech signal separation and speech transformation using neural network.

    本文介紹作者在進行說話人識別、混疊語音信號分離和應用神經網路技術進行語音轉換方面的若干研究探索問題。
  12. In the thesis, we select the mel - frequency cepstrum coefficients based on analyzing a lot of parameters of speech signal. mel cepstrum is of better recognition and anti - noise capability. ( 2 ) dynamic time warping, vector quantization, hidden markov model and artificial neural network can be used in speaker recognition

    ( 2 )現有的說話人識別方法有動態時間規整法、矢量量化法、隱馬爾可夫模型和神經網路法等,其中hmm已成為目前最佳的說話人識別處理模型。
  13. Hands - free speech communication is indispensable in audio and video conference systems, hot - line telephones and videophones, mobile radio terminals, digital isdn network etc. however, the control ( cancellation ) of the acoustic echo has always had a strong impact on the transmission quality in hands - free telecommunication ~ [ 1 - 3 ] conventional methods of acoustic echo control ( cancellation ), such as echo suppression or gain control, may lead to the degradations in speech quality or make the speakers feel uncomfortable

    免提式話音通信在移動電話、熱線電話、車載電話及isdn網的電視電話會議等多種領域正得到日益廣泛的應用。但至今的免提式話音通信中,仍免不了受回聲引起的話音失真、甚至嘯叫等干擾,大大降低了免提話音通信的質量。由於聲回波對消問題尚未得到圓滿解決,實現高質量的免提式話音通信仍是一個極賦挑戰性的課題。
  14. The softswitch technology breaks the close traditional exchange structure. it adopts the combined pattern, open interfaces and generally used agreements to establish a open nad distributed system structure for more customers ' application. the substance of softswitch is to turn traditional switch matrix to the ip network. the network is open, easy and cheap. it makes use of superiorities of the network to exchange speech sounds

    軟交換技術打破了傳統的封閉交換結構,採用組合模式、開放的介面和通用的協議,構成一個開放的、分佈的和多廠家應用的系統結構。軟交換的實質是將傳統交換的電路矩陣移到ip網路上實現。利用ip網路的開放、簡單、低廉等優勢進行語音交換。
  15. Tcs can be used as the basic software system for www and telecommunication value - added service dealer, furthermore it can also regard as management entity of h. 323 speech classroom network

    Tcs可以作為門戶網站、電信增值運營商的基本軟體系統,也可以理解為h . 323語音教室網路的管理實體。它是所有h . 323網路內呼叫的焦點。
  16. This article mainly discusses the basic theories and related protocol and technology of speech sound communication based on ip. the key discuss is the support of real - time services that is provided by the current ip network, codec and the acoustic - echo cancellation, as well as the dynamic monitor of network communication quality of service. it gains some conclusion by compare the merit and shortcoming between the h. 323 protocol and sip protocol and analyzing their bases : rtp / rtcp protocol

    本文主要論述基於ip的語音通信所涉及的基本理論和相關的協議與技術,重點論述當前ip數據網路對實時業務提供的支持、語音編碼、回聲消除和網路通信服務質量的動態監測,在協議的分析上比較h . 323和sip協議之間的優缺點,並具體分析這兩個協議的基礎: rtp rtcp協議。
  17. Synthesis of prosodic information for chinese mandarin text - to - speech using a wavelet neural network

    漢語文語轉換系統中基於小波神經網路的韻律信息合成
  18. Design and layout of a digital network speech classroom

    數字網路語音室的設計與布設
  19. The g729a, a new issued speech compressed coding standard at 8kbit / s, is one of the optional coding standards of h. 323, which is multimedia meeting system standard based on ip network

    729a是國際電信聯盟新頒布的編碼速率為8kb s的低速率語音壓縮編碼標準,它是基於ip網路的多媒體會議系統標準h 323可選的語音壓縮編碼標準之一。
  20. So we must think out an effective way to settle them as quickly as possible, in this article, i put forward my scheme including image, speech, network and transmitting of file and so on to settle these problem. there are a lot of knowledge have been included in dealing with image ; they are image collection, image saving, image color distill ing, palette dealing with, image compression, image packing, image compounding, image improving etc. in dealing with speeh aspect, speech collection, speech compression, speech recording, speech playing have been involved.,

    在本方案中涉及到圖像、語音、網路、文件傳輸等方面的知識。其中的圖像部分涉及到圖像的採集、保存、顏色的提取、調色板的處理、壓縮、打包、合成、增強等方面的知識;語音方面涉及到語音的採集、保存、壓縮、錄取、回放等知識;網路方面涉及到基本的網路傳輸協議、組播、 winsock編程、網路阻塞事件的解決以及網路的監控等方面的知識;文件傳輸方面涉及到文件的分發、上傳、下載、刪除以及文件的傳輸協議ftp等知識。
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