語聲信號 的英文怎麼說

中文拼音 [yǔshēngxìnháo]
語聲信號 英文
voice signal
  • 語 : 語動詞[書面語] （告訴） tell; inform
  • 號 : 號Ⅰ名1 （名稱） name 2 （別號; 字） assumed name; alternative name3 （商店） business house 4 （...
  1. The results of simulation indicated speech signal processed by the optimum algorithm presents obvious periodicity in time domain, and effect of the formant is removed or restrained effectively in frequency domain

    處理后的語音信號在時域上表現出明顯的周期性特徵，同時在頻域上也觀察到聲道的共振峰結構影響得到消除或有效的抑制。
  2. Based on the speech produce model, we find the reason of periodicity disappearance and the extremum number increase by analysing the character of speech signal when the glottal closes

    於是從語音產生模型入手，詳細的分析了聲門閉合時刻語音信號的性質，找到了濁音信號經過小波變換後周期性消失、極值點個數增多的原因。
  3. If the wavelet transform is directly implemented in pitch detection, comparing the glottal closure singularity of speech signal with image grey break, we will not obtain the anticipative result

    將聲門閉合在語音信號中表現出相應的奇異性，與圖像邊緣的灰階突變進行等價對比，直接將小波變換用於聲門閉合奇異型的檢測，並不會得到預期效果。
  4. The appropriate tune for fire information display interface was mezzo - soprano

    在消防監控界面中，語音信號的適宜語聲為女中音。
  5. Because the speech signal is periodicity at sonant which vocal cords surge in low frequency and similarity to white noises at surd, the pitch can be detected in traditional way through the correlation operation without the speech produce model

    在人類語音的濁音段，聲帶發生較低頻率的振蕩，語音信號呈明顯的準周期性，而在清音段，語音信號則類似於白噪聲。
  6. In accordance with chaotic essence of speech signals, syllable segmentation in continuous speech is researched by fractal theory. an approach of syllable segmentation using variance fractal dimension is proposed, its performance is analyzed in detail. the method can discriminate between voiced and unvoiced, between surd and sonant, but it can hardly discriminate between sonant

    本文根據語音信號的混沌本質，利用分形理論研究了漢語連續語音中的音節分割問題，提出了基於方差分形維數的音節分割方法，並詳細分析了該方法的性能，它能很好地解決了無聲與有聲、濁音與清音間的分割問題，但很難解決濁音間的分割問題，當濁音相連且過渡段較短時，該方法無法實現它們之間的分割。
  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. Theoretical expatiate on general concepts and fundamental principles of information hiding and steganography, also point out possible directions for further research, also analysis the probability of speech as the host carry signal and efficient masking characteristics of psycho - acoustic model, it is shown that : there is an improvement on imperceptibility according to human auditory masking effect

    闡述了信息隱藏技術和隱寫技術的重要概念、基本理論以及廣闊的應用前景。分析了將語音信號作為宿主載體信號的可行性，參考心理聲學模型的特性，得出結論:基於人耳聽覺掩蔽效應的隱寫演算法，在隱蔽性上有很大的提高。
  9. The result of experiments show that resynthesized speech signals form the its correlogram by auditory model inversion is nature and robust in noisy environment

    實驗結果表明，我們通過聽覺模型反演從信號的自相關圖譜中恢復出的語音信號，具有較好的自然度和良好的噪聲魯棒性。
  10. The international speech communication association covers all the aspects of speech communication ( acoustics, phonetics, phonology, linguistics, natural language processing, artificial intelligence, cognitive science, signal processing, pattern recognition, etc

    國際言語溝通學會：涵括了言語溝通所有面向的主題（聲學，語音學，音韻學，語言學，自然語言處理，人工智慧，認知科學，語言信號處理，型樣識別等） 。
  11. 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。
  12. The statistic of wavelet transform coefficient algorithm can solve the periodic noise, high - energy noise and some non - gauss noise simply and effectively ; bi - spectrum can acquire more information from the original signal than power - spectrum, detect more information except from range and restrain the gauss noise. short - time speech signal can be considered as stationary and with periodic non - gauss signal, so we can make use of bi - spectrum to obtain the speech character and separate the speech and noise and detect morse telegraph signal ; complex number spectrum variance algorithm is put forward based on the deeply observing speech data, it is a new algorithm, experiment show that it is simple, effective

    統計演算法在解決周期信號、高能噪聲和高斯信號方面有獨特之處，能簡單有效提取以上噪聲的特徵;雙譜能夠提供比功率譜更多的有用信息，有效地檢測信號幅度之外的其它信息，並能有效抑制高斯噪聲，短時語音信號一般認為是平穩且有一定的周期性的非高斯信號，因而可以利用雙譜來提取語音信號特性並實現信噪分離;復數譜方差演算法是在對語音信號進行深入觀察和分析的基礎上而提出來的一種全新的語音特徵提取方法，此方法簡單而有效的提取了語音、噪聲的特徵以及檢測莫爾斯信號，基於實驗表明，該演算法取得了很好的效果。
  13. First, the traditional speech detection method based on short - time energy is discussed, including its principle and implementation. then it is used for the jumping - off point detection of speech signals transmitted by awgn channel. simulation results are provided

    包括該方法的原理、實現，並將其應用於加性高斯白噪聲通道干擾下的語音信號起點檢測實驗，給出了實驗的統計結果。
  14. The traditional detection algorithm, based on zero - crossing or energy, will not acquire ideal effect when the signal - to - noise is low or the signal is weaker. therefore, to resolve the real problem in the real environment that all kinds of random noise and speech signal exit together, some new algorithm must be put forward. account for the complexity of real noise, we integrate the wavelet transform and high - order statistics and advance a new algorithm ; the algorithm can effectively separate the speech signal and the non - gauss noise

    基於過零率和能量的傳統檢測演算法，在噪聲環境比較復雜的情況下效果很不穩定，尤其是信噪比較低或者語音信號較弱時，檢測效果很不理想，因此，在多種語言和噪聲隨機出現、噪聲和語音強弱不一的實際噪聲環境下，必須利用新的演算法提取有用信號和噪聲信號的有效特徵，才能解決實際的問題。
  15. The demand is the power forcing speech coding to progress. traditionally linear prediction ( lpc ) vocoders are very efficient, which can encode speech from 800 to 2400bps, but unfortunately, artifacts such as buzzes, thump, and tonal noise always exist in them

    經典的線性預測（ lpc ）聲碼器具有很高的編碼效率，可以極低的碼率（ 800 2400bps ）對語音信號進行編碼，不幸的是它的合成語音聽起來很不自然，常常夾雜著嗡嗡聲，重擊聲或者音調噪聲。
  16. Spectrum analysis of vehicle noise and speech signal and the application of a digital subsidence filter

    汽車噪聲和語音信號的譜分析及陷波器的應用
  17. The matching pursuit techniques are applied to enhance speech signal, and a method to determine the threshold of coherent ratio is provided in the enhancement procedure based on matching pursuit. with the method, the noisy signal can be efficiently enhanced in a rather wide range while the statistical property of signal and noise is unknown

    運用匹配跟蹤技術處理了語音信號增強問題，給出了匹配跟蹤信號增強過程中相干比閾值的確定方法，實現了在未知信號與噪聲統計特性的情況下，在相當大的范圍內明顯增強信號的目的。
  18. In this paper, we use full pole model to obtain speech signal lpc, then deduce it ' s lpcc, and we use the lpcc difference to describe speaker ' s track dynamic movement

    本文應用全極點模型，提取語音信號的線性預測系數，並推導出其倒譜系數，獲得線性預測倒譜差分，用以描述說話人聲道的動態變化。
  19. We made an improvement in overcoming the defects in speech signal adaptive delta modulation ( abbr. adm ), such as slope overloading and grain noise. in this method, numerical sliding average filtering was used for filtering decoding speech signal. experiments and analyses indicate that the method makes waveforms in good agreement between the decoding of adm and the original pulse coding modulation ( abbr. pcm ) signal, and considerably improves, the playback speech quality in naturalness, legibility and under standability

    針對語音信號自適應增量調制（ adm ）方式中斜率過載和顆粒噪聲缺點，提出了一種改進方法，它利用滑動平均方法對解碼后的信號進行數字濾波.試驗和分析表明，該方法使解碼后的信號波形與原脈沖編碼調制（ pcm ）波形具有很好的一致性，使再生語音質量在自然度、清晰度和可懂度方面比改進前均有較大提高
  20. The article researches on the application of adaptive filtering theory on the noise processing of speech signal based on the characteristics analysis of speech signal

    本文在對語音信號進行特性分析的基礎上，主要研究了自適應濾波理論在語音信號噪聲處理中的應用。
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