語音信號 的英文怎麼說

中文拼音 [yǔyīnxìnháo]
語音信號 英文
speech signal
  • 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
  • 號 : 號Ⅰ名1 （名稱） name 2 （別號; 字） assumed name; alternative name3 （商店） business house 4 （...
  • 語音 : speech sounds; pronunciation; voice
  1. An agonic algorithm of speech logarithm spectrum envelope

    一種語音信號對數幅度譜包絡的無偏演算法
  2. Autocorrelation algorithm for speech pitch detection based on matlab

    的語音信號自相關基音檢測
  3. The hardware of the ip phone codec to be designed is based on the fixed point digital signal processor ( ti ' s tms320vc5410 ) while the compression and decompression core in the software of dsp is based on the itu - t vg. 729a. ip phone codec carryout the task of collecting / playing - back. coding / decoding of speech signal and communication with embedded cpu. etc

    該語音編解碼器的硬體基於tms320vc5410 ，編解碼演算法遵循itu - tg . 729a協議，能夠實現語音信號的採集/回放、編碼/解碼以及同嵌入式cpu通信等功能，在8kbit / s的碼率下能夠提供獲得良好的語音質量。
  4. 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。
  5. 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。
  6. 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。
  7. 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。
  8. 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。
  9. 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。
  10. In hyper dimensional phase space, similar sequence repeatability ( rpt ) of speech are calculated

    研究語音的相似序列重復度及其熵信息，分析比較了語音信號在相空間中的非線性特徵。
  11. 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。
  12. 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。
  13. After about two years " insisting and hard working, this goal set at the beginning has become true. the developed c54x general assembly program for g. 729 speech signal compressing algorithm has passed the tracking with more than 3, 000 unitary standard measuring vectors. g. 729 speech signal compressing compiler using c54x general assembly program has been accomplished real - timely, and undistorted rebuilt speech signals have been obtained

    因此本課題選用c54x的通用匯編語言編程實現g . 729語音壓縮編碼演算法，調試並通過了統一標準測試矢量三千多幀，最終在5402開發實驗板上實時實現了g . 729語音壓縮編碼器，獲得未失真的重建語音信號。
  14. 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。
  15. 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。
  16. 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。
  17. With the auditory model as the front - end to extract the correlogram of signals. following, this paper present the implementation of suditory model inversion procedure by resynthesizing original signal from the correlogra - m

    接著，文章闡述了通過實現聽覺模型反演過程從信號的自相關圖譜中恢復出原始的語音信號的過程。
  18. In this paper, on the basis of absorption of achievements of the research on auditory physiology, an auditory model simulationg the peripheral auditory system and part of the central auditory system is set up. the model is made of the fitlters presenting the characteristics of the basilar membrane for analyzing the voice signals, the half wave rectification modeling the inner hair cells and energy transfer of nerve fiber

    在吸收聽覺生理學研究成果基礎上，建立了一個模擬外圍聽覺系統和部分中樞聖經系統功能的聽覺模型。模型由表徵基底膜的頻率分析的帶通濾波器組、內毛細胞的半波整流特性和神經纖維的能量轉換特性組成，該模型可以作為前端處理來提取語音信號的自相關圖譜。
  19. The thesis makes researches on technologies of call centers, voice disposing technologies and the synthetic application of these technologies. with setting up a call center which is called epcc ( electric power call center ) in a electric power company, the thesis describes pivotal technologies in call centers, such as voice disposing technologies >, crm ( customer relationship management ) and web technologies, the thesis represents standard schemes and standard frames of call centers, and the thesis describes pivotal technologies in voice disposing procedures, such as speech synthesize and speech recognize, and the thesis describes voice disposing technology " s applications in call centers that are called ivr ( interactive voice response ) systems, then the thesis discusses the acd ( auto call distribution ) program in epcc

    本論文通過建立呼叫中心的一個實例（電力呼叫中心） ，對呼叫中心、語音技術及其綜合應用進行了較為深入的研究。通過呼叫中心的電力應用，較詳細地論述了呼叫中心的關鍵技術：語音技術、客戶關系管理crm和web技術等；較深入地闡述了呼叫中心的典型方案和典型結構；較詳細地論述了語音信號處理技術的關鍵技術：語音合成技術及語音識別技術：深入討論了語音技術在呼叫中心的具體應用- - - ivr系統及其關鍵技術；較詳細地討論了電力呼叫中心中所採用的acd演算法，並基於acd演算法完善了呼叫中心的表現形式。
  20. It is conceived to introduce barker codes as synchronization preamble and add synchronization signal in front of speech signal to implement speech detection. the principle of this idea is presented

    提出了引入barker碼作為同步碼，在語音信號前添加同步信號以實現起點檢測的工作設想，並介紹了其同步原理。
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