語音通信系統 的英文怎麼說
中文拼音 [yǔyīntōngxìnxìtǒng]
語音通信系統
英文
vcs voice communications system- 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
- 通 : 通量詞(用於動作)
- 系 : 系動詞(打結; 扣) tie; fasten; do up; button up
- 統 : Ⅰ名詞1 (事物間連續的關系) interconnected system 2 (衣服等的筒狀部分) any tube shaped part of ...
- 語音 : speech sounds; pronunciation; voice
- 通信 : communication; communicate by letter; correspond
- 系統 : 1. (按一定關系組成的同類事物) system 2. (有條理的;有系統的) systematic
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This thesis tries to update the cmdsr system to achieve the characters below : real - time, better robust, higher recognition rate, non - special - man. considering the disadvantages of traditional improved spectrum subtraction speech enhancement, this thesis proposes the theory of fuzzy spectrum subtraction based on the fuzzy theory and improved spectrum subtraction speech enhancement ; as for the difficulties of detecting the endpoint of speech signal, the thesis gives the table of initial and the improved parameters, with which we can confirm the endpoints of mandarin digit speech ; the thesis puts forward two - level digit real - time speech recognition system, the first level is based on discrete hidden markov model which is linear predictive coding cepstrum ( lpcc ) and difference linear predictive coding cepstrum ( dlpcc ), the second level is based on formant parameters ; as for the realization of hardware, the thesis depicts the realization of every part of cmdsr based on the tms320vc5402 in detail ; as for the development of software, the thesis gives the software design flow chart of cmdsr, simulates the basic theory with matlab language and gives the simulation results
針對傳統的「改進譜相減法語音增強」參數設定單一、環境適應能力差的缺點,提出了一種利用模糊理論和「改進的譜相減法」結合的「模糊譜相減法語音增強」 ;針對語音信號端點檢測困難的特點,通過matlab模擬試驗,給出了能夠準確確定數碼語音端點的初始和改進參數表;提出了利用基於線性預測編碼倒譜參數和差分線性預測編碼倒譜參數相結合的離散隱含馬爾可夫模型進行第一級識別、利用共振峰參數進行第二級識別的兩級漢語數碼語音識別系統,在保證系統實時性的同時,實現連接漢語數碼語音識別系統識別率的提高;在硬體實現上,詳細闡述了基於tms320vc5402的連接漢語數碼語音識別系統各部分硬體設計;在軟體開發上,給出了連接漢語數碼語音識別的軟體設計各部分的流程圖,並對各部分進行了matlab模擬,並給出了模擬結果。Speech recognition is a cross - disciplinary, is gradually becoming it, the key to human interface technology, voice recognition and voice synthesis technologies combining technology integral voice response systems, voice synthetic systems, interactive toys, enable people to voice ordering through operation, the control mode flexibility, a broad market prospect
語音識別是一門交叉學科,正逐步成為信息技術中人機介面的關鍵技術,語音識別技術與語音合成技術相結合組成語音應答系統、語音合成系統、互動式玩具等,使人們能夠通過語音命令進行操作,控制方式靈活方便,具有廣闊的市場前景。It is expected to be used for 3g personal handy - phone system as standard algorithms which encode speech signals and decode it. additionally, this kind of algorithms which own excellent quality can be application in viewphone and video order programme etc. the thesis introducethe algorithm structure of g. 729
該協議在可預見的將來可能應用於三代移動通信系統中作為語音編解碼演算法。另外,由於其良好的性能也可應用在多媒體系統中如:可視電話,視頻點播等。本論文概要介紹了g . 729協議的演算法結構。Ip voice gateway is a node between the communication control unit and the server of trunking system in trunking system network. its duty is digital compress coding and packetization of the voice in trunking system. it located on the edge of the network and accessed the trunking system into ip network from the kernel network angle
語音網關作為集群通信系統ip聯網的關鍵設備,它處于集群系統話路控制單元和集群系統服務器之間,起著語音壓縮、分組傳輸的重要作用;從網路的角度來看,它處于ip網路和集群無線通信系統之間,把集群通信系統接入ip網路。So acoustic echo cancellation ( aec ) technique has become a hotspot of competition in famous communicaton company all over the world. however, it is not always easy to implement acoustic equipments for communication system with satisfactory speech quality
回波消除技術能有效地解決長距離電話網路、 ip電話、免提電話和視頻會議等通信系統中的回波問題,很好地改善了語音通信質量,具有廣闊的市場前景。Then this spectral subtraction method is applied to noise speech recognition system as the front - end processing. noise speech signal are processed to improve its snr before recognition. so the recognition rate can be improved in noise environments
並將改進譜減演算法作為噪聲下語音識別系統的前端處理過程,即通過對含噪的語音進行語音增強以提高信號的信噪比,從而提高語音識別系統的抗噪聲性能。Speech communication is one of the most used modes in the digital trunking communication system. excellent algorithm of speech coding can save the bandwidth resource, improve the utilization of frequency, so it has important value for investigation
語音通信是數字集群通信系統中最常用的通信方式之一,優良的語音編解碼演算法能夠更加有效地節省帶寬資源,提高頻率利用率,因此具有重要的研究價值。After implementing digital signaling system into analogue trunking system, we get these improvements : more reliable connection between users, higher speed of connection, and more functions in trunking system. mdc1200 signaling system is one kind of this signaling, which is developed by motorola for improving the performance of its analogue trunking system
模擬集群通信是指它採用模擬語音進行通信,整個系統內沒有數字技術,但后來為了使通信連接更為可靠,模擬集群通信系統也採用了數字信令,使集群通信系統的用戶連接比較可靠,連通的速度有所提高,而且系統功能也相應增多。The main functions of the system are implemented, including browsing digital map, navigation and positioning, communicating, and connecting internet. the results are very well
本系統可以實現的主要功能有:電子地圖瀏覽、 gps定位導航、短消息通信、語音通信、 gprs上網、最短路徑搜索、中文輸入等。Allowing uavs to fly alongside airliners will require them to develop the means to “ sense and avoid ” other planes ; new air - traffic control systems, based on electronic rather than voice communications, will also be needed
允許無人機與各航班的飛機在同一片天空飛行就需要這些無人機能具有「感應與避免」其它飛機的功能;而基於電子通信而不是語音通信的新型空運控制系統同樣也將成為必需。The algorithm have the good one - way property, high sensitivity to initial values and good security due to the intrinsic characteristic of chaotic system and rijndael algorithm. the simulation experiment demonstrates the convenience and good hash performance ; 3 ) a new scheme of digital voice secure communication was proposed based on chaotic modulation without additional synchronization. the modulation sequence generated by chaotic logical mapping was used to encrypt the digital voice signal
混沌系統和rijndael演算法的固有特點使該演算法具有較好的安全性、對初值有高度的敏感性以及較好的單向性能,並且易於實現,是一種有效的單向hash函數; 3 )研究了一種無需同步的基於混沌調制的數字語音保密通信系統的方案,利用邏輯映射產生混沌調制序列,以該序列作為密碼對數字語音進行加密處理,為了更好的隱匿信號特徵,混沌調制在小波分解的基礎上,對不同的通道使用不同的參數進行,並借鑒混沌掩蓋對信息信號進行了限幅處理,使密文完全隨機化。Specifically, this thesis includes the following work : the basic theory of speech processing and noise in speech reduce. we study the short interval character of speech signal and short interval process technology, and classify the noise and disturbance in wireless receiver terminal into 3 categories and present corresponding schemes
本文針對高質量無線語音傳輸的需求,對語音處理及語音降噪的基礎理論、關鍵技術及降噪設備的實現方法展開研究,在此基礎上設計實現了無線語音通信降噪器實驗系統,並對實驗系統的降噪效果進行了測試。A voice system should ensure high quality voice first. meanwhile, the net bandwidth is very finite and precious resource
語音通信系統首先必須保證良好的話音質量,而網路帶寬又是非常有限和寶貴的資源。A secure speech communication system based on a digital chaos
一類數字混沌保密語音通信系統This thesis makes a research on the design and implementation of an underwater digital voice communications system, which can realize the high speed transmission of digital voice streams in real time and implement voice communications with a good quality
本論文對水下數字語音通信系統的設計和實現進行了研究,開發了一套水下數字語音通信系統,該系統實時的實現水下數字語音的高速傳輸,並能完成高質量的語音通信。Design of wireless digital audio communication system and its application in field medicine
無線語音通信系統的研製及其在野戰醫療中的應用3. finish demonstrating, designing and realizing an underwater voice communication system. code, modulation, equalization and demodulation are realized in real - time
對水下語音通信系統進行方案設計論證,完成編碼調制、均衡和解調在硬體系統上的實時實現。Voip transmits voice over an open network, and the voice data includes a great deal of important information, so it needs much more securities than the traditional data communications
網路語音通信系統需要考慮的另外一個因素是其安全性。網路語音通信是在開放的網路上進行的,因此,它比傳統的語音通信更容易受到攻擊。This thesis mainly deals with the hardware design of whole system, on the basis of summarization of underwater voice communication systems, this thesis analyzes some key technology used in underwater digital voice communications, demonstrates the feasibility of the whole system, and then provides a scheme to realize the whole system
本論文以整個系統的硬體設計為主要研究對象,在綜述水下語音通信系統的基礎上,首先分析了水下數字語音通信中的幾項關鍵技術,論證其可行性,並由此提出了整個系統的實現方案。The paper emphasizes on the discussion of the problems in implementation of network speech communication system of pc to pc in lan and the schema of secure network speech communication. the aes algorithm, the chaotic sequence and the method of chaos modulation was adopted to encrypt the voice signal and the srp protocol was used for identity authentication, so as to strengthen security. and some pivotal problems is discussed and settled, such as the compressing, restraining of reverberation and the obliteration of voice signal dither
著重考慮了局域網環境下的pc到pc的網路語音通信系統的實現的相關問題;探討並設計了網路語音傳輸中的保密解決方案,採用aes演算法、混沌序列密碼加密方法和混沌調制方法對語音武漢理工大學碩士學位論文信號進行加密處理,選用srp協議用於身份認證,以提高網路語音通信的保密性能;並針對網路語音傳輸中的幾個關鍵問題,包括語音信號的壓縮、網路語音信號的靜音抑止和網路語音的去抖動處理等問題進行了分析。分享友人