speech noise 中文意思是什麼

speech noise 解釋
言語噪聲
  • 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。
  • noise : n 1 聲音，聲響。2 叫喊；嘈雜聲，噪音；喧鬧聲；吵鬧，騷動，騷擾。3 〈古語〉謠言，風聲。4 〈美國〉...
  1. After realized the arithmetic, add the complexity, such as add sonority noise, paroxysmal noise, etc. than used the method of amending parameters, reference cyclostyle and two stage recognition to improve the precision of speech recognition

    Dsp方案採用tms320vc5409一80為核心，搭配ad轉換器tlc2558 、電源晶元tps73hd318等組成完整的硬體電路。
  2. Moreover in speech enhancement, especially in reducing the pulse noise, morphological algorithm has its unique advantage. particularly morphological filter may maintain the preferable accurate of the speech signal in speech waveform, and which produces little impairment to the formant of speech. so the spectrum structure of the speech is retained well, and the quality of the speech will not be reduced

    特別是，在時域波形分析中，形態學濾波增強較小波去噪更好地保持語音信號的細節；在頻域分析中，形態學濾波對語音信號的基音頻率、頻譜斜率、共振峰等語音特徵的影響很小，因而能夠較好的保留語音信號的頻譜結構，使語音品質不致降低。
  3. It reduces the “ music noise ” using the human auditory characteristics, and enhances the hearing quality using the speech spectrum distribution characteristics in the time - frequency dimension

    主要以譜減法為基礎，結合人耳的聽覺特性從而減少殘留「音樂噪聲」的影響;結合語音的語譜在時-頻域分佈特性從而提高增強后語音的聽覺質量。
  4. According to the different characteristics between signal and noise on wavelet transform domain, also considering the voiced and unvoiced speech has different features, a modified method of speech denoising which is using a changing threshold at different scales is proposed

    摘要分析了信號和噪聲在小波域的不同特徵表現，並根據語音中濁音和清音的特點，提出了一種改進的多尺度多閾值的小波域語音去噪方法。
  5. That is, using a soft thresholding to remove noise components from the wavelet coefficients of the voiced and unvoiced speech in noisy speech respectively in a different way, which is not only removing noise but also is preventing the quality degradation of the unvoiced sounds and enhancing the signal - noise ratio

    該方法採用軟限幅函數對濁音和清音信號的小波變換系數作不同的閾值處理，既抑制了噪聲，又減少了語音段信息的損失，提高了信噪比。
  6. Experimental results in different noises and snr indicated that this vad algorithm can divide speech segments from non - speech segments accurately and reduce voiced - unvoiced error obviously. ( 2 ) an improved dct - hn speech decomposition algorithm based on the harmonic - noise model is presented

    不同噪聲、信噪比下的實驗結果表明，該演算法可以準確區分語音段與非語音段，明顯降低了基音檢測中清濁誤判現象的發生; （ 2 ）基於「諧波-噪聲」模型提出了一種改進的dct - hn語音分解演算法。
  7. Annex b introduce a voice activity decision ( vad ) algorithm which class speech signal as voice signal and background noise signal

    Annexb提出了一種靜音壓縮演算法（ vad ） ，它將語音信號分為話音信號和背景噪聲信號。
  8. Suart has looked at many of the cartoon greats tintin, asterix, calvin and hobbes, the moomins and little nemo in slumberland and has used many of the standard devices of the language of cartooning ; the speech and think bubble, the materialised noise, the half - face close - up, the fisheye lens view, the silhouette, multiple perspectives, the nimbus round the head to signify presence or surprise

    創作中，小話參照了多個經典卡通人物，如tintin 、 asterix 、 calvin與hobbes 、 moomins以及slumberland里的little nemo ，同時運用很多卡通的標準技法，如說話泡泡或思考泡泡、圖像化的聲音、半邊臉大特寫、魚眼鏡頭角度、人物體態、多種透視法、表示驚訝或神怪的頭上光環等。
  9. In order to reduce the musical residual noise and the background noise, a speech enhancement method based on masking properties of the human auditory system is described. this method uses bark wavelet packet transform to simulate the frequency feature of human auditory model to get the threshold

    本文以最大限度減少殘留噪聲和背景噪聲為目的，採用bark子波分析的方法模擬人耳基底膜的頻率分析特性來進行語音增強，重點進行模擬人耳聽覺掩蔽效應來確定除噪閾值的研究。
  10. 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
  11. 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。
  12. According to the project of adaptive multi - rate speech coding ( amr ) being put forward by the third generation group of the mobile communication, this paper takes the principle of the speech arithmetic as the base, studies the technologies including the source controlled rate, voice activity detector, comfort noise and the error concealment unit in amr, discusses its the characteristic of adaptation and analyses its performances particularly. amr c codes are researched carefully through the modules being divided into and debugged under the tms320c54x provided by the ti corporation, and optimized in selecting the method of c code embedded assembler codes and simplified in the search codebook combining with the theory of speech coding, which are based on the realization about theory and practice of the optimization of amr speech coding

    從自適應多碼率語音編碼演算法的c代碼出發，對它進行模塊劃分後作了系統分析，將其在vc下調試通過，進一步在ti公司提供的tms320c54x環境中進行調試，結合語音編碼理論，對演算法進行優化，採用了在c代碼中嵌入匯編和簡化自適應碼本和固定碼本搜索的方法，部分地提高了c代碼效率，為實現自適應多碼率語音編碼的優化奠定了理論和實踐基礎。
  13. Speech enhancement as the front - end processing module is used to improve the signal - to - noise ratio ( snr ) of the input signal for recognition in the latter stages

    為了讓語音識別系統在安靜的環境和有噪聲的環境中都獲得令人滿意的工作性能，研究了一個將語音增強器和語音識別器級連起來的系統。
  14. 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。
  15. Because the adaptive algorithm of conventional adaptive noise canceller is the least mean squares ( lms ), and the convergence rate of lms is heavily dependent on the eigenvalue distribution of the autocorrelation matrix of the input signal, thus lms converges at unacceptably low rates when the input signal is colored noise or speech

    由於傳統自適應噪聲抵消系統（ anc ）自適應演算法主要採用lms演算法，而lms演算法收斂速度依賴于輸入信號自相關矩陣特徵值的分散程度。因此，當輸入信號是語音或有色噪聲時， lms的收斂速度很慢。
  16. Computer simulation results show that the proposed algorithm reduces the signal distortion and the reverberation which are caused by misadjustment errors in the adaptive filter and the correlated component of the speech in the reference signal compared with a conventional adaptive noise canceller ( anc )

    模擬結果證明nanc演算法有效克服了影響傳統自適應噪聲抵消系統（ anc ）性能的一些瓶頸，如兩輸入中的非相關噪聲，參考輸入中的信號成分與自適應濾波器失調誤差而產生的信號失真、回響等情況。
  17. Dsp technologies for reducing noise in speech. we study the character of dsp and discuss the procedure to design noise reducing system, and discuss some issues for system architecture, design and debug. design and implementation of the prototype of a wireless speech noise reducing device

    第三，本文研究了語音降噪系統的硬體技術，介紹了dsp晶元的構成和特點，討論了基於dsp晶元構造降噪系統的設計流程，研究了系統構成、系統設計和系統調試的關鍵技術。
  18. This dissertation is different from traditional speech enhancement methods which are based on noise characteristic such as adaptive noise cancellation or spectral subtraction processing. in this dissertation the speech signal conducted by bone was taken as the object to be studied and the exploitive study on the acoustical characteristic of speech signal conducted by bone was performed by the method of theory combined with experiment. then a proposition about speech reconstruction based on speech signal conducted by bone was presented, and the design of software and hardware was completed

    本文與傳統的基於噪聲特性的自適應噪聲抵消法、頻譜減法等語音增強降噪技術不同，是以骨導語言為研究對象，採用理論與實驗相結合的方法對骨導信號的聲學特性進行了探索性研究，進而提出了基於骨導信號的語音重構技術，並完成了相應的軟硬體開發。
  19. Speech enhancement method based on masking properties of the human auditory system is used to reduce the white noise in the front - end

    摘要為了提高噪聲環境下說話人識別系統的識別性能，將基於聽覺掩蔽效應的語音增強技術作為預處理器，對語音信號首先進行降噪處理，提高輸入信號的信噪比。
  20. As the pitch doubling and halving problems of nccf algorithm often occurred with varied noises and signal to noise ratio ( snr ), vad algorithm is employed to separate speech and non - speech segments

    針對nccf基音檢測演算法在不同噪聲、信噪比下容易發生清濁誤判的問題，本文在基音檢測前端引入語音檢測演算法劃分語音段與非語音段。
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