線性預測系數 的英文怎麼說

中文拼音 [xiànxìngyùcèxìshǔ]
線性預測系數 英文
linear prediction coefficients
  • 線 : 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...
  • 性 : Ⅰ名詞1 （性格） nature; character; disposition 2 （性能; 性質） property; quality 3 （性別） sex ...
  • 預 : Ⅰ副詞（預先; 事先） in advance; beforehand Ⅱ動詞（參與） take part in
  • 測 : 動詞1. （測量） survey; fathom; measure 2. （測度; 推測） conjecture; infer
  • 系 : 系動詞（打結; 扣） tie; fasten; do up; button up
  • 數 : 數副詞（屢次） frequently; repeatedly
  • 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
  • 系數 : [數學] coefficient; ratio; modulus; quotient; factor
  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。
  2. In order to utilize the frquency resource adequately and increase the capacity of mobile communication system, the wireless electric wave propagation of existing mobile system always adopts the microcell structure. forecasting the path loss characteristic of electric wave accurately can provide the necessary condition for the layout and design of wireless network, at the same time it is a precondition for the research on the microcell mobile system. the methods of forecasting of wireless electric wave propagation can divide into two parts : one is pluse and respond, that is establish the empirical model based on experimental and statistical data ; the other is ray tracing method, that is establish the deterministic model based on theoretical analyse. the paper discuss the characteristic of wireless signal electric wave transmition in symmetrical atmosphere of earth, and introduce the common path loss transmition model in land mobile communication system, also point out the localization of these models based on experiential methods

    而精確預測無線電波傳播路徑損耗特性，則為合理的微蜂窩無線網路規劃、設計提供了必要條件，同時也是研究微蜂窩移動通信系統性能的前提。無線電波傳播預測的方法分為兩類：一是用沖激響應法，即根據實驗、統計所得數據建立經驗性傳播預測模型；另一種是用射線跟蹤方法，即依據理論分析來建立確定性的傳播預測模型。本文首先討論了在地球表面均勻大氣中的無線電波傳播的基本特性，介紹了陸地移動通信系統中常用的幾種電波傳播路徑損耗經驗性預測模型，並指出了這些經驗性傳播模型對于微蜂窩小區無線電波傳播特性研究的局限性。
  3. Carry on emulation to melp standard, realize that the compression of the pronunciation file is solved and pressed. first this thesis sample to wav file, carry on the speech to analyze and draws with the parameter to the speech data of every frame. these parameter include pitch, bpvc, jitter, lpc, etc. then, these parameters will be quantized by msvq technology

    該系統首先對語音信號進行采樣；按幀對語音數據進行語音分析和參數提取，提取的參數包括基音周期（ pitch ） 、多帶清濁音判別、非周期抖動標志、線性預測參數（ lpc ）等語音生成模型參數；接著對這些參數進行了量化，量化採用了多級矢量量化技術；最後在解碼端對各個量化參數進行解碼，利用這些參數結合語音合成模型重構語音。
  4. The approaches establish a relationship between monthly precipitation abnormality and monthly circulation, soil moisture and temperature on the shallow and deep layers. the relationship is the precipitation diagnostic equation and its coefficients and dimensions are determined by using the observed data of huai river basin. then we select the main soil moisture and temperature attributing factors by the dimensional analysis to establish a forecasting equation of summer precipitation over huai river basin with the statistic approach

    通過將大氣中的熱量、水汽收支方程與一個簡化的兩層土壤溫度、濕度方程相結合，並依據月尺度大氣環流的演變特徵，推導出月降水距平與500hp月平均高度距平場、土壤深淺兩層溫、濕度的關系；利用臺站觀測資料，使用統計反演方法確定方程中各項的系數和量級，從而找出影響降水的主要土壤溫、濕因子；利用統計方法建立這些因子與淮河流域夏季降水異常之間的簡單線性預報方程，並對1992 - 2000年淮河流域夏季降水趨勢進行回報。
  5. Wavelet coefficients are encoded by the arithmetic encoder, with the contexts being formed by quantizing linear prediction values

    它通過量化當前系數的線性預測值形成上下文，把系數作為一個整體進行自適應的算術編碼。
  6. 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性，採用美爾倒譜系數識別密碼，線性預測倒譜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。
  7. At the same time, built the data warehouse system with the sale subjects as the example, built an many - dimensions database using the online analysis tool of microsoft sql sever. finally built a two linear forecasting models based on smoothing of time queue about analysis of sale trend, and verified the design analysis of this paper partly

    同時，建立了以商品銷售主題為例的數據倉庫系統；並藉助microsoftsqlsever聯機分析工具，建立了以商品分析主題為例的多維數據庫，從不同視角展現不同匯總程度的數據；最後，建立了基於時間序列的二次指數平滑線性預測模型，進行商品銷售趨勢的分析，部分驗證了本文的設計分析。
  8. By means of the precise integration method with lagrangian interpolation the trajectory of the shaft center, the poincare mapping and the bifurcation graphs are numerically given. the results predicted by the floquet theory are checked and the long - term dynamic behavior of the system is predicted. it is shown that the system has rich nonlinear behaviors at some m combination of the four parameters, for examples, multi - frequency subharmonic resonance, as well as chaos phenomenon from doubling bifurcation and twice hopf bifurcation

    通過lagrange插值精細積分法數值給出系統的軸心軌跡圖、 poincar映射圖、分叉圖，檢驗floquet理論預測結果並預測系統的長期性態，顯示系統在四個參數組合的某些范圍內具有豐富的非線性特性，還存在多形式次諧波解，以及由倍周期分叉、二次hopf分叉通往混沌的現象。
  9. Tetra - the new digital trunked communication system, specifies algebraic codebook excited linear prediction ( acelp ), which is improved codebook excited linear prediction ( celp ), as its speech codec. the bit rate of it is 4. 567kbit / s

    本文首先詳細介紹了新一代數字集群通信系統tetra的語音編解碼採用的編碼速率4 . 567kbit s的代數碼本激勵線性預測（ acelp ）演算法。
  10. A neural network method for generrating the linear prediction coefficients of random sequences

    一種求線性預測系數的神經網路方法
  11. 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。
  12. 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

    由於未量化的線性預測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，具有與語音信號譜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。
  13. Second, a novel algorithm named model predicition ( mp ) is proposed to wipe off spectral correlations of hyperspectral images. mp algorithm finds the linear model of hyperspectral images, in which predictive coefficients are set up that is based on snr. because predictive coefficients include current spectral band, average entropy of the error data is decreased and snr is increased after mp

    Mp演算法建立了高光譜圖像的線性模型，推導出了信噪比意義下的最佳預測系數，由於系數中包含了當前譜帶的數據，因此經過mp演算法去相關之後，殘差圖像的平均熵有所降低，同時信噪比提高很多。
  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維線性預測系數和12維線性預測倒譜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. With the flying development of voica synthesis technique, desiging voice synthesis device at low price is in the face, therefore we devise voice respinsive system which uses coice synthesis processor as core chip in this paper and can turn numeric information into voice export using linear forecast coding technique, and we gain satisfying effect

    隨著語音合成技術的飛速發展，設計低價格的語音合成裝置已迫在眉睫，因此本文設計了以語音合成處理器為核心晶元的語音應答系統，我們利用線性預測編碼技術把數字信息變成語音輸出，獲得了滿意的效果。
  16. 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、付立葉譜特徵參數、功率譜特徵參數、時域振幅特徵參數、線性預測編碼系數、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預測需要。
  17. Discussion on the factors of controlling the distribution of the reservoir quot; sweet spots quot; of sulige gasfield

    數控伺服進給系統中的非線性預測控制
  18. Speech enhancement is becoming an important branch of speech signal process, which mainly used in the noise reduction, the preprocess of the speech recognition system and lpc

    語音增強目前已發展成為語音信號數字處理的一個重要分支。它的主要應用是降低聽覺噪聲、作為識別系統的預處理和線性預測編碼的預處理。
  19. Meanwhile, the telephone gateway in tetra system is introduced. in further research, the principle of tetra speech coding algorithm ? algebraic codebook excitation linear prediction ( acelp ) is introduced and analysed in detail, which is a advanced codebook excitation linear prediction ( celp ). acelp algorithm replaces the excitation signals with algebraic codebook and uses some technique such as minimizing the mean square error ( mse ) and the analysis - synthesis method to obtain characteristic parameters of speech

    同時，介紹tetra系統的市話網關，並在接下來的研究中詳細介紹tetra電話網關中應用到的語音編解碼演算法? ?代數碼本激勵線性預測碼（ acelp ）的基本原理，它是一種簡化了的碼本激勵線性預測碼（ celp ） ，它把激勵信號用代數碼本代替，並且運用了均方誤差最小、分析?合成等技術提取出語音的特徵參數，極大地降低了比特率，而且具有較好的重建語音質量。
  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預加重和加窗分幀處理等，由niosii軟核執行端點檢測、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線性預測美爾倒譜系數（ lpcmcc ） ，採用動態時間規整（ dtw ）的識別演算法。
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