高音譜號 的英文怎麼說

中文拼音 [gāoyīnpǔháo]
高音譜號 英文
g clef, treble clef
  • 高 : Ⅰ形容詞1 （從下向上距離大; 離地面遠） tall; high 2 （在一般標準或平均程度之上; 等級在上的） above...
  • 音 : 名詞1. （聲音） sound 2. （消息） news; tidings 3. [物理學] （音質） tone 4. （姓氏） a surname
  • 譜 : 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）...
  • 號 : 號Ⅰ名1 （名稱） name 2 （別號; 字） assumed name; alternative name3 （商店） business house 4 （...
  • 高音 : [音樂] high pitch; high-pitched voice
  1. In piano music the upper staff usually has a treble clef and the lower a bass clef, the former usually for the right hand and the latter for the left in music for the piano, or for similar instruments

    在鋼琴作品中，上面的五線譜通常標有高音譜號，下面的五線譜標有低音譜號，在鋼琴作品或類似樂器作品中，前者通常為右手彈奏，後者為左手彈奏。
  2. 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。
  3. At their music stands the orchestra were tuning their instruments amid a delicate trilling of flutes, a stifled tooting of horns, a singing of violin notes, which floated forth amid the increasing uproar of voices

    樂師們對著樂譜架調整樂器的音色，笛子的輕快顫音，法國號的低沉呼鳴，小提琴的悅耳奏音交織在一起，在越來越高的嘈雜人聲上空蕩漾。
  4. Then this spectral subtraction method is applied to noise speech recognition system as the front - end processing. noise speech signal are processed to improve its snr before recognition. so the recognition rate can be improved in noise environments

    並將改進譜減演算法作為噪聲下語音識別系統的前端處理過程，即通過對含噪的語音進行語音增強以提高信號的信噪比，從而提高語音識別系統的抗噪聲性能。
  5. 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。
  6. 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。
  7. Super resolution pitch detection based on band - partitioning spectral entropy and signal decomposition in dct domain

    分帶譜熵與信號分解的高精度基音檢測演算法
  8. One is using the autocorrelation function to detect the speech terminal, the other is using the coefficients based on the one - sided autocorrelation sequence to replace the original speech signal and then extract the speech feature to recognize. isolated word recognition experiment based on dtw shows : it can reduce the disturbance of noise effectively and get the high recognition rate. it is of great advantage to apply when snr signal to noise rate is low

    對含噪語音在自相關域上進行處理，以其自相關函數值為參數進行端點檢測，以基於單邊自相關序列的lpc倒譜系數作為語音的特徵參數進行語音識別，實驗表明：這種方法較好地消除了噪聲對語音信號的干擾，並獲得了較高的識別率。
  9. Dynamic mapping algorithm is also illustrated in details. through the computer simulation to some real short - time voice signal samples using matlab language. the result shows that the recognition efficiency using cepstrum coefficients mapping is better than what made by lpc mapping

    實驗結果表明，與採用lpc特徵相比，採用lpc倒譜特徵和動態匹配演算法進行短時語音識別，會有較高的識別率；對不同語音信號有特徵空間離散度大、易於確定判別門限的特點。
  10. Point on the is again and again the s high segment is with of the low segment at oppositely loudly the degree objective relation for exsitting of ; also point a voiceses the way the way of high fidelity and left voice similar to signal for the way s of right voice of. equilibrium

    指在音頻頻譜的高段和低段之間在相對響度上所存在的客觀關系也指雙聲道立體聲左聲道和右聲道之間的信號的相同平衡。
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