重音變換 的英文怎麼說

中文拼音 [zhòngyīnbiànhuàn]
重音變換 英文
stress modification
  • 重 : 重Ⅰ名詞（重量; 分量） weight Ⅱ動詞（重視） lay [place put] stress on; place value upon; attach im...
  • 音 : 名詞1. （聲音） sound 2. （消息） news; tidings 3. [物理學] （音質） tone 4. （姓氏） a surname
  • 換 : 動詞1. （給人東西同時從他那裡取得別的東西） exchange; barter; trade 2. （變換; 更換） change 3. （兌換） exchange; cash
  • 重音 : 1. [語言學] stress; accent 2. [音樂] accent; 重音符號 stress mark; accent; 重音節 stressed syllable
  1. Can use stress patterns, words in stress, rhythmic structures, and intonation contours in order to make the utterance fit for the aim, motivation, attitudes, state of emotions, etc

    能夠運用重音、邏輯重音、節奏和各種語調的變換來適應溝通目的、動機、態度以及情緒情感的變化。主要是通過各種特定形式來表達不同的態度和情感。
  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. At last the audio - visual filter can get and regroup the pcm audio data which transformed into the time - frequency signals by the fft. at the same time the power spectrum can be displaying on the windows by using the win32 api functions. finally the test of this system shows the processes about transport stream of receiving in the collecting card, pushing in the bridge, separating in the parser, displaying in the audio - visual filter are satisfied with the data rata at the server

    音視頻效果filter內部通過獲取解碼后的pcm格式數據，並利用緩沖管理器對這些數據進行重新分組，然後對分組后的數據進行fft變換將時域信號變換到頻域之中，同時對變換后的數據作數學處理后得到其功率值，最後利用api畫圖函數對這些功率值進行了實時的特殊效果顯示。
  4. It is followed by digital sampling, windowing, noise filtering, endpoint detection, time - domain vector and transform - domain vector

    內容涉及語音信號的數字化、加窗處理、預加重濾波、端點檢測，及時域特徵向量和變換域特徵向量。
  5. This paper introduced that how resynthesize the neural firing rate by correlogram inversion, how resume the lost information by half wave rectification inversion. and a solution of speech enhancement based on masking properties of human auditory is proposed and implemented, we can not only analyse and resynthesize speech signal, but also analyse and resynthesize speech signal in noisy entironment with the system

    介紹了如何從信號的自相關圖譜中逆變換中得到信號的神經發放率函數，怎樣從信號的半波整流逆變換中恢復丟失的信息等等。又根據聽覺生理特點提出了抗噪聲方案，使我們的語音分析重構系統不僅可以對無噪信號進行重構，而且還可以在噪聲環境下的語音信號進行重構。
  6. Vorbis coding algorithm can provide high quality audio compression with advanced perceptual coding and transform coding technologies ; and because of its completely open and non - patent, vobis has attracted more and more attentions by music industrial and audio users

    Vorbis編解碼演算法利用先進的感知編碼和變換編碼等技術提供高質量的音樂壓縮，同時因為其完全開源和無專利版權費用的特點而逐漸受到音樂工業和用戶的重視。
  7. In this paper, we briefly introduced the performance of wave coding and vocoder, emphasizedly studied the principle and performance of variable rate vocoder q4401, including the internal construction and pins, qcelp coder & vocoder, pcm interface, cpu interface initialization process, command format and so on. we also designed a application circuit, with the experiment validated its performance. in this design, the pcm interface chip is tp3057, it was used to finish a / d transform, the compress coding was finished by q4401, the initialization and control were accomplished by 8051 singlechip

    重點是研究變速率語音編解碼晶元q4401的工作原理及性能。其中包括q4401的內部結構及管腳、 qcelp編碼方式、 pcm介面、 cpu介面、初始化過程、命令格式等，並在此基礎上，設計一個實際的應用電路，通過實驗，驗證其性能。在設計中用pcm介面晶元tp3057來完成從模擬信號到數字信號的轉換，再由q4401進行壓縮編碼，對q4401的初始化及控制由8051單片機來完成。
  8. We bring forward the method through compare the two projects adaptive spectrum enhancement and wavelet analysis which combines the wavelet analysis and the adaptive theory and have the information of several frequence bands being the reference input signals to reach the adaptive processing and restructure of speech signal by making use of merits of the two theories

    通過對比自適應譜線增強和小波變換的兩種方案，提出了把小波分析理論與自適應理論相結合，利用兩種理論的優勢，用小波分解后的若干層頻段信息作為參考輸入信號，實現語音信號噪聲的自適應處理和重構。
  9. 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。
  10. The author had done a lot of reading work on the evolution of circuit switch phone network to packet switch phone network and acquainted with the three main branches of vop technology. by focusing on voip the author gained the basic knowledge of h. 323, sip and h. 248 / megaco architechture and then can study the h. 323 series protocol. because the project refers to a network management module the author also studied the most popular network management technologies

    作者查閱了大量相關資料，了解了電話系統從電路交換到分組交換的演變過程，以及分組語音的三個主要分支的形成與發展，進而專注于其中最典型也最具前途的ip語音技術，了解了h . 323 、 sip及h . 248 / megaco系統的原理，並重點學習了h . 323系列協議。
  11. Subsequently, the various audio and video compression protocols in surveillant system, especially the fundamentals of wavelet compression technology are introduced in chapter two. the comparison between wavelet transform and dct transform is given. the schemes of intra - frame coding and sequence frame coding based on wavelet compression are used as a standard in current system

    第二章介紹了監控系統中所採用的各種音視頻壓縮協議，對小波壓縮技術的一些重要概念作了簡要的介紹，詳細比較了小波變換和dct變換的優缺點，給出了基於小波壓縮的單幀及序列幀編碼方案，並將其作為本系統的視頻壓縮標準。
  12. The peak components in the pre segmentation image and the distance - transformation image were extracted by area reconstruction dome improved transform, and were fused to form markers image

    基於面積重構頂改進變換濾除噪音能力強的特點，分別提取預分割圖和距離灰度圖中的峰值區域，融合後作為分水嶺改進變換的標識，由此得到分水線，從而完成對黏連氣泡的分割。
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