語音變換 的英文怎麼說
中文拼音 [yǔyīnbiànhuàn]
語音變換
英文
phonetic modification-
The preordained frangibility of the hymen, the presupposed intangibility of the thing in itself : the incongruity and disproportion between the selfprolonging tension of the thing proposed to be done and the self abbreviating relaxation of the thing done : the fallaciously inferred debility of the female, the muscularity of the male : the variations of ethical codes : the natural grammatical transition by inversion involving no alteration of sense of an aorist preterite proposition parsed as masculine subject, monosyllabic onomatopic transitive verb with direct feminine object from the active voice into its correlative aorist preterite proposition parsed as feminine subject, auxiliary verb and quasimonosyllabic onomatopic past participle with complementary masculine agent in the passive voice : the continued product of seminators by generation : the continual production of semen by distillation : the futility of triumph or protest or vindication : the inanity of extolled virtue : the lethargy of nescient matter : the apathy of the stars
女性之虛弱及男性之強韌乃基於謬誤的臆測。道德的準則是可變的。自然的語法轉換：在不引起意思變動的情況下，由主動語態不定過去式命題從語法上分析：男性主語，單音節擬聲及物動詞，女性直接賓語轉位到相關的被動語態不定過去式命題： 3 」從語法上分析：女性主語，助動詞與準單音節擬聲過去分詞，男性主動補語。Speech enhancement based on discrete cosine transform
基於離散餘弦變換的語音增強A novel speaker normalization method based on formant recovery and mellin transform
基於子波變換的自適應濾波語音增強方法In various speech character parameters, formant frequency, bandwidth and pitch frequency are chosen as voice character parameters. the reasons are as follows : hearing apperceive experiments indicates that formant frequency can stand for a majority of voice information, while average pitch frequency can explain 55 % ability of speaker verification
數據結果與多項式回歸和線性多變量回歸相比，支持向量回歸既提高了泛化性能又避免了頻譜不連續性，從而使轉換語音與目標語音的頻譜距離失真分別減少了33 . 29 %和35 . 24 % 。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。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。Algorithm for pitch detection of chinese tri - syllabic words based on wavelet transform
基於小波變換的漢語三字詞語音基音頻率提取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。Fast wavelet analysis algorithm based on oblique projection and mellin transform
基於小波包變換的說話人語音特徵參數的提取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。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。Lpc prediction error ; one - side autocorrelation sequence lpc ; acoustic front end processing ; canonical correlation based on compensation ; combination of features
線性預測誤差單邊自相關線性預測語音前端聲學處理正則相關分析的譜變換補償特徵綜合The main aim of speech signal processing is speech coding, speech recognition and speech understanding by automaton. in this paper, firstly, some basic signal process are discussed, such as signal filter, sampling, fft, pitch detection, then, they are tested on the testing board designed by the author
論文中首先對語音信號的基本處理問題進行了分析和對比，然後在自己設計的基於tms320vc5402的dsp實際系統上，進行了語音處理過程的濾波、采樣、傅立葉變換和譜包絡提取的演算法實現研究，討論了在演算法的dsp實現方法，分析了運行實驗結果。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。Computer used in telecommunication not only play as control part in exchange or transmission devices, but also become a part of speech communication to provide intelligentized speech service
計算機已經不只是簡單作為電信交換和傳輸網路的控制部分而存在，而是逐漸演變成語音通信的一部分，將智能化的語音延伸到通信網路的每一個角落。A speech enhancement method using adapted filter based on wavelet transform
基於小波變換系數自適應閾值法在語音去噪中的應用Speaker normalization and novel robust speech feature based on mellin transform
變換的語音新特徵與頻率歸正說話人自適應技術Then based on it an audio encryption algorithm is proposed by adopting module operation, which has nice encryption characters in the simulation. another audio encryption algorithm based on zero dynamical nonlinear invertible system is studied and improved. in order to compare the encrypiton characters between the improved and the original algorithm, their applications to audio encryption are simulated, and the corresponding results are shown in chapter 3
本文的主要成果有:提出了一種混沌實值序列的生成方法，並在此基礎上引入模變換對語音信號進行加密，經實驗模擬可以得到良好的加密效果;對零動態可逆混沌語音加密通用模型的一個具體加密演算法進行了改進，並通過模擬實驗比較了改進前後演算法的加密性能。There are difficulties in noisy speech recognition, especially low signal - to - noise rations are more difficult. this paper describes briefly six methods for speaker - dependent noisy speech recognition isolated words. they are lpc prediction error method, one - side auto - correlation sequence lpc, acoustic front end processing, canonical correlation based on compensation method, combination of features method and increase of poles method. the experimental results show that all the six techniques can improve effectively noisy speech recognition, and the best noisy speech recognition rate is above 80 % when snr 0db
它們是：線性預測誤差法，單邊自相關線性預測法，語音前端聲學處理法，正則相關分析的譜變換補償方法，特徵綜合法和同模極點增加法。實驗結果表明，這6種方法都有效地提高了噪聲環境中語音識別率，其中較好的方法在強噪聲環境中信噪比為0db的語音識別率達到80 %以上，為信噪比較低的噪聲環境中自動語音識別展現了美好前景。Comparing to polynomial regression and linear multi - variant regression, support vector regression can not only enhance the generalization ability but also avoid the discontinuousness in spectrum, and the spectrum distance distortion from converted voice to target voice are reduced to 33. 29 % and 35. 24 % respectively
與多項式回歸和線性多變量回歸相比，支持向量回歸既提高了泛化能力又避免了頻譜不連續性，使轉換語音與目標語音的頻譜距離失真分別減少了33 . 29 %和35 . 24 % 。分享友人