語音理解系統 的英文怎麼說

中文拼音 [yǔyīnlǐjiěxìtǒng]
語音理解系統 英文
speech understanding 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
  • 理 : Ⅰ名詞1 （物質組織的條紋） texture; grain （in wood skin etc ） 2 （道理;事理） reason; logic; tru...
  • 解 : 解動詞（解送） send under guard
  • 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
  • 理解 : understand; comprehend
  • 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
  1. On the basis of the study on the speech coder algorithms, paper describe an advanced method of developing dsp system software, and as the guidlines, we developed the programme of whole decoder unit. paper stress on analysis of the ecu in decoder unit. aiming at amr algorithms disadvantage of angularity of synthetical speech, paper study on the specutral extrapolation which apply to extrapolate reflect coefficient of track model to make error conceal processing of amr. at last paper analyze existing echo cancellation algorithms using on mobile communication system

    在此基礎上，描述了一種較為先進的大型dsp系統程序開發策略，並以此為指導思想，以美國ti公司c6000dsp開發平臺開發出了整個amr解碼器單元的系統程序。論文對amr解碼器的誤碼隱藏處理單元進行了重點分析，針對原有演算法合成語音自然度不好的缺點，論文研究了將譜外推法應用到amr演算法中外推出聲道模型反射系數參數進行誤碼消除處理。
  2. The system is consist of the main data processing board which is based onthe fpga device and fast ethernet phyceiver rtl8201l and a - law pcm data encoder and decorder chip msm7702 - 3, and the dial - up and display board which is based on mcu. the main board would carry out the core task of data processing, such as voice data packing and unpacking, the ethernet frame processing, protocol processing, call processing, etc. the dial - up and display board would carry out the task of display the ip address which is input by consumer and status of network during talk period from the main board, and so on. in the paper the system of lan ip telephone and the tcp / ip protocol is introduced firstly, then the fpga device is stated. after that the fpga - based hardware scheme is introduced in detail in chapter four

    系統以altera公司的acex1k系列的fpga和快速以太網控制器rtl8201l和語音編解碼晶元msm7702 - 3為核心構建了數據處理主板和以單片機為控制器的撥號顯示子板組成。數據處理主板的核心任務，包括語音數據處理、以太網幀處理、協議處理、呼叫處理等。撥號顯示子板則完成通話前的顯示用戶所撥過的ip地址，通話期間網路狀態的顯示等等。
  3. For the real time performance need of the low speed speech compress algorithm and the asic implement of the transfer process between programs, the design is put forward in the paper, in which state registers control the cross access between operator and memory, register windows are used for the parameters transfer, and the technique of hardware controlling is used to avoid pipeline conflict, so that the main problems of the transfer process in tr600 are solved effectively

    摘要針對低速率語音壓縮演算法對處理器系統實時處理復雜運算的性能要求，就程序調用過程的asic實現問題進行了對比與分析，進而提出了用層次狀態寄存器控制存取運算元對存儲體交叉訪問的方法，並結合運用寄存器窗口傳遞參數的功能，以及利用空指令硬布線處理流水線沖突的方法，有效地解決了tr600晶元中調用過程存在的主要問題。
  4. Speech coding systems are generally based on narrowband speech at present, but its frequency is restricted in 200hz ~ 3400hz and its sample rate is 8khz. with the development of the wideband speech, its bandwidth which is from 50hz to 7khz causes the quality of speech communication to approach in the feeling of face - to - face conversation, and makes the speech natural, expressive and comfortable. hence, it ’ s quite significant in researching on wideband speech coding system. in recent 20 years, the dsp and its software development kit has improved greatly, but the price has fallen sharply, thus it has more and more widespread applications now

    隨著當今世界的飛速發展，寬帶語音越來越受到人們的青睞，因為它的50hz 7khz的帶寬使得語音通訊質量接近於面對面交流的感覺，大大提高了語音的自然度、表現力和舒適度。因此，開發研製基於寬帶語音的編解碼系統具有十分重要的意義。在過去的短短二十年裡， dsp處理器的性能得到很大的改善，軟體和開發工具也得到相應的發展，價格卻大幅度地下降，從而得到越來越廣泛的應用。
  5. 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。
  6. Under the condition of " comparatively weak correlation between the two noises involved, coherence function is used as a frequency domain amplification factor for improving snr of the output signal to the filter and the speech enhancement effect. meanwhile, a real - time recursive algorithm is put forward in substitute for current algorithms based on short time fourier transform. the new algorithm will simplify computations and will be suited for real - time implementation together with the adaptive systems

    接著針對上述nanc系統兩路輸入信號噪聲相關性弱的情況，用相干函數作頻域增益因子來提高輸出信噪比與改善語音增強效果，同時，通過一種實時迭代演算法解決了短時傅氏變換計算量大的問題，簡化了計算，便於實時處理與實際應用。
  7. The system will display the gate status, the thunder status and fire signal status. it allows the management clearly receiving running status of all facilities and subsystems in the central control room. the system graphically displays the alarm position, informs management by phonetic alarm and provides relevant disposal method for the management, which reduce cost of undesired contingency

    虹口稅務局基於節能和高效管理的考慮，在夜間，如果雙鑒探頭檢測到有人活動，則自動控制燈光開啟與關閉，同時聯動視頻進行錄像，並在界面上實時反映門的開關狀態防雷的狀態消防信號的狀態等，讓機房管理人員通過機房監控系統統能夠全面地了解機房內所有設備和子系統的運行狀態，並形象地顯示報警發生的位置，進行語音提示，同時提示相應報警的處理方法，將損失減少到最小。
  8. However, few people actually understand precisely what the term means, what the consequences of a particular damping factor rating might be for the ultimate desired result ( sound reproduction

    然而，只有很少人真正能夠理解這個術語的準確含義，了解某一特定的阻尼因子量度如何能對最終聲音重放的結果產生影響，或者有那些因素和系統或阻尼因子有關。
  9. 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。
  10. Those communication methods used have advantages and disadvantages in those aspects, such as communication quality, adaptation condition, covering scope, geography environment, construction and working cost, real time quality, etc. in order to make the transport of hydrological information better, this dissertation introduced gsm communication methods in the hydrology telemetric system, using gprs cell - phone, by those mobile communication methods, such as gsm sms, gprs sms, gprs data service, circuit data service, tone channel, etc. this dissertation summarized the constitute of hydrology telemetric system, mobile communication methods, and compared them with those traditional methods

    現有的水情遙測通信方式有多種類型，它們在通信質量、適應條件、覆蓋范圍、地理環境、建設成本及運行費用、實時性等方面各有優缺點。為了更好地進行水情信息傳輸，本文提出一種基於gsm通信方式的水情遙測系統，通過使用gprs蜂窩電話，採用gsm短消息、 gprs短消息、 gprs數據業務、電路型數據業務和語音通道等多種移動通信方式，為水情數據的傳輸提供了新的解決途徑。
  11. 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。
  12. The pulsar system is developed with the following parts : the robust feature extraction based on auditory mechanisms, acoustic modeling with mandarin pronunciation characteristics, search space building based on wfst, and efficient global decoding scheme

    本研究實現的系統pulsar包括：基於聽感知機理的抗噪聲特徵提取，針對漢語發音特點的聲學建模，基於加權有限狀態自動機（ wfst ）的搜索空間，融合聲學模型和語言模型的高效全局解碼。
  13. The operating system and control program run on host processor mpc852t, the voice compress codec program run on the voice processor ac48304, they communication with the host port interface

    主處理器mpc852t上運行操作系統及控製程序，語音壓縮分組處理器ac48304進行語音壓縮編解碼。
  14. In the paper, the embeded process platform and voice compression moudle were introduced firstly, then host port interface circuit was designed by which the host processor mpc852t communicated with ac48304. and then em interface and ethernet interface of the gateway were designed, which communicatewith trunking system and server respectively

    本文的主要工作是在了解嵌入式處理平臺和語音壓縮模塊的基礎上，設計了powerpc處理器擴展ac48304的hpi介面電路;並設計了語音網關與集群系統的em介面以及以太網介面。
  15. The service system discussed in this paper and the custom system compose the whole 112 system, which provides a flexible solution for the operator ’ s malfunction management

    本文所實現的業務系統與語音客服系統一起構成的112系統，為電信運營中的故障管理提供了完整靈活的解決方案。
  16. 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。
  17. The paper expatiates design of the practice apply system using the ambe arithmetic, and introduces the correlation field and applied range about voice ' s digital signal process. and talks about the superiority of low speed voice ' s vocoder

    本論文主要研究了應用ambe演算法，進行實際的應用系統設計，介紹了語音數字信號處理涉及到的研究領域，應用范圍，語音高效編解碼低速率的優勢。
  18. Speech understanding system

    語音理解系統
  19. Meanwhile, the telephone gateway in tetra system is introduced. in further research, the principle of tetra speech coding algorithm ? algebraic codebook excitation linear prediction ( acelp ) is introduced and analysed in detail, which is a advanced codebook excitation linear prediction ( celp ). acelp algorithm replaces the excitation signals with algebraic codebook and uses some technique such as minimizing the mean square error ( mse ) and the analysis - synthesis method to obtain characteristic parameters of speech

    同時，介紹tetra系統的市話網關，並在接下來的研究中詳細介紹tetra電話網關中應用到的語音編解碼演算法? ?代數碼本激勵線性預測碼（ acelp ）的基本原理，它是一種簡化了的碼本激勵線性預測碼（ celp ） ，它把激勵信號用代數碼本代替，並且運用了均方誤差最小、分析?合成等技術提取出語音的特徵參數，極大地降低了比特率，而且具有較好的重建語音質量。
  20. 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。
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