線性預測譜 的英文怎麼說

中文拼音 [xiànxìng]
線性預測譜 英文
linear prediction spectrum
  • : 名詞1 (用絲、棉、金屬等製成的細長的東西) thread; string; wire 2 [數學] (一個點任意移動所構成的...
  • : Ⅰ名詞1 (性格) nature; character; disposition 2 (性能; 性質) property; quality 3 (性別) sex ...
  • : Ⅰ副詞(預先; 事先) in advance; beforehand Ⅱ動詞(參與) take part in
  • : 動詞1. (測量) survey; fathom; measure 2. (測度; 推測) conjecture; infer
  • : Ⅰ名詞[書面語]1 (按類別或系統編成的書或冊子等) table; chart; register 2 (指導練習的格式或圖形)...
  • 線性 : [數學] [物理學] linear; linearity線性代數 linear algebra; 線性方程 linear equation; 線性規劃 line...
  • 預測 : calculate; forecast; prognosis; divine; forecasting; foreshadowing; predetermination
  1. 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「改進的相減法」結合的「模糊相減法語音增強」 ;針對語音信號端點檢困難的特點,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;提出了利用基於編碼倒參數和差分編碼倒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。
  2. For a set of gasoline samples, multivariate linear regression ( mlr ) and partial least squares ( pls ) calibration models are built to predict research octane number ( ron )

    針對一批實際生產裝置的汽油樣品,採用近紅外光定量分析中常用的多元回歸( mlr )與偏最小二乘( pls )方法,建立了汽油研究法辛烷值nir光模型。
  3. One - sided autocorrelation sequence ; linear predictive coding ; cepstrum ; dynamic time warping

    單邊自相關序列編碼倒動態時間規正
  4. Linear predictive coding ; lpc prediction error ; cepstrum ; dynamic time warping

    編碼lpc lpcpe誤差倒動態時軸彎曲或動態時間規正dtw
  5. And its conversion of line spectral pair ( lsp ) and the production of adaptive codebook are different from that of traditional celp

    對( lsp )的轉換和自適應碼本生成方面也採用了有別于傳統碼本激勵演算法的新技術。
  6. On the basis of analysis and comparison between two drills, one in the center of bohai sea, another near the west shore of bohai sea, ultra - long electromagnetic wave remote sensing can be applied to forecast the interfaces between the different rocks, and help to choose the location of drill and drilling plan. the ultra - long electromagnetic remote sensing also can be applied to general investigation in the prospecting area and organizing the structural map on the basis of the profiles and plane. based on the analysis of the ultra - long electromagnetic wave curves from tanggu to dalian, the geological body to effect the high gravity and magnetic anomalies could be a mafic intrusion. the magma activity provided the heat source to organic maturation in the center of bohai sea, so the center of bohai sea could be the prospection of deep gas in bohai sea

    根據渤海西岸和渤海中部兩口探井的探和對比實驗分析,利用超長電磁波遙技術可以根據已知探井的探對比分析新探井的巖界面,協助井位的選址和設計。另外,利用超長電磁波的探技術可以從剖面和平面上對遠景區進行普查,編制遠景區的構造圖。根據塘沽-大連探的超長電磁波頻剖面對比分析,證實引起渤海中部重磁異常高的地質體可能是基超基巖體。
  7. Lpc prediction error ; one - side autocorrelation sequence lpc ; acoustic front end processing ; canonical correlation based on compensation ; combination of features

    誤差單邊自相關語音前端聲學處理正則相關分析的變換補償特徵綜合
  8. Considering system security, we adopt mfcc to recognize password and lpcc to represent speaker track dynamic movement. the double decrees enable it applying in high secret situations. the system has many merit such as the quick operation velocity, easy model update, less calculate quantity and low error rate

    本文考慮到系統的安全,採用美爾倒系數識別密碼,差分識別說話人聲道動態變化的雙重判決方法,為系統應用在高度機密場合提供了可能,具有運算速度快,模板更新容易,計算量小,差錯率低等優點。
  9. Each band of hyperspectral image has the same physical structure, so we classification the first band, and design an optimal linear predictor for each class to make the mean prediction square error minimal, and then we use jpeg - ls algorithm to remove the spatial redundancy

    由於高光圖像每個波段都具有相同的物理結構,先對首幅圖像進行分類,在每個子類中分別使用各自的最佳器,將該類中的相鄰段進行並將殘差均方降為最小,然後用jpeg - ls演算法去除殘差圖像的相關
  10. Secondly, hyperspectral images are hard to compress because of their abundant details, complicated texture and insignificant special correlation. making use of the significant spectral correlation within the hyperspectral images, we propose an optimal linear predictor which makes the square error minimal

    針對高光遙感圖像細節豐富紋理復雜,空間相關弱,難于壓縮的特點,本文充分利用了高光遙感圖像的間相關,設計出對相鄰段進行並將殘差均方降為最小的一種最佳器。
  11. Linear prediction code and mel - scale cepstrum coefficients are effective method in speech feature extraction and important in speech signal processing

    分析和mel倒分析是對語音信號特徵提取比較有效的兩種方法,在語音信號處理中有著重要的應用。
  12. 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,提取語音信號的系數,並推導出其倒系數,獲得差分,用以描述說話人聲道的動態變化。
  13. The quantized lp coefficients are replaced by the unquantized lp coefficients in the frequency domain expression of the feel weighted filter. the error signal has more similar envelope shape, and the hearing effect is better than before because the unquantized lp coefficients have more accuracy than quantized lp coefficients

    由於未量化的系數具有更高的精度,因此,誤差信號通過修正後的感覺加權濾波器以後,具有與語音信號更加相似的包絡形狀,從而更好地利用共振峰對誤差的掩蔽效應,達到更佳的主觀聽覺效果。
  14. In the phase of training, it gets the sampling data from the wave files which were stored in the voice library by using the mci functions. then calculates the character vector ( 12 ranks of lpc and lpcc ) and trains them by clustering method, so we get the templates used by speech - recognition, this templates were stored in the template library. in the state of recognition, after calculating the character vector of input voice, we compare it with the character vectors of templates, and then find the best one or refuse it

    系統的組成模塊與語音識別系統的基本構成模型基本一致,在訓練過程中,通過調用mci ( mcimultimediacontrolinterface )提供的函數從語音庫中的波形文件中讀取采樣數據,分幀計算出由12維系數和12維系數構成的特徵矢量,並按照聚類的方法進行訓練,得到后續語音識別時需要的模板,存放于模板庫中。
  15. 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:誤差法,單邊自相關法,語音前端聲學處理法,正則相關分析的變換補償方法,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。
  16. Based on high - dimension space geometry, every speech sample is looked as a point in space. then the speech sample point is extracted feature by lpc, mel - scaled cepstrum analysis or auto correlation - angle. their feature is looked as a point too

    基於高維空間幾何的思想,把每個樣本點和其特徵值看作高維空間中的一個點,用分析、 mel倒分析和自相關夾角法對樣本點提取特徵,然後用點在空間的投影來判別語音和非語音,根據判別結果來比較三種特徵提取方法的優劣。
  17. These are correlation characteristic parameter, fourier spectrum characteristic parameter, power spectrum characteristic parameter, time domain amplitude characteristic parameter, linear prediction coding coefficients, instantaneous characteristic parameter, absorb and decay coefficient, velocity characteristic parameter and wavelet packet transform characteristic parameter. the parameters contain the surface relatively wide, the prediction which is suitable for the goal of many kinds of seism needs

    分別為:自相關特徵參數、付立葉特徵參數、功率特徵參數、時域振幅特徵參數、編碼系數、瞬時特徵參數、吸收衰減系數、速度類特徵參數和小波包變換特徵參數,參數涵蓋面較寬,適用於多種地質目標的需要。
  18. The theory of lpc and mel - scaled cepstrum analysis is introduced in this dissertation and how to extract lpcc and mfcc is elaborated

    詳細介紹了分析( lpc )和mel倒分析的原理及其具體實現過程。
  19. So, in this paper, the theory and algorithm of vr are being developed. in this paper, several key problems in vr process are being discussed both in theory and application, which include pre - processing, frame decomposing of raw voice signal, characteristic selection and calculation, dynamic mapping of characteristics. linear prediction model, model coefficients ( lpc ), as well as cepstrum coefficients are well analyzed both in analysis and calculation aspects

    作者在本論文中,對國內外語音識別技術發展狀況做了較全面的總結分析,對語音信號產生模型、編碼方法、求解lpc正則方程的德賓遞推演算法、語音信號同態處理方法、 lpc倒特徵計算、動態特徵匹配等語音識別的關鍵環節的技術問題進行了深入的理論分析和模擬研究,用matlab語言編寫了語音信號濾波、分幀、特徵計算和匹配軟體,並給出了模擬計算結果。
  20. Nios ii soft core fulfills endpoint detection, feature extraction, discipline, recognition, input control and output display, etc. the audio signal feature, in this scheme, is the lpc mel cepstrum coefficient ( lpcmcc ) and recognition algorithm is dynamic time warping ( dtw )

    由fpga硬體完成對音頻數據的加重和加窗分幀處理等,由niosii軟核執行端點檢、特徵提取、訓練建模、識別匹配、輸入控制和輸出顯示等。系統提取的音頻信號特徵為美爾倒系數( lpcmcc ) ,採用動態時間規整( dtw )的識別演算法。
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