語噪比 的英文怎麼說

中文拼音 [yǔzàobǐ]
語噪比 英文
speech to noise ratio
  • 語 : 語動詞[書面語] （告訴） tell; inform
  • 噪 : 動詞1. （蟲或鳥叫） chirp 2. （大聲叫嚷） make noise; make an uproar; clamour
  • 比 : Ⅰ動詞1 （比較; 較量高下、 長短、距離、好壞等） compare; compete; contrast; match; emulate 2 （比...
  1. Focused on the application of the generalized cross correlation ( gcc ) time delay estimation ( tde ) in the car microphone array denoising system, gcc tde algorithm is analyzed theoretically and compared with high rank cumulation tde

    摘要針對汽車環境中麥克風陣列語音去噪系統的應用，對廣義相關時延估計方法進行了模擬，並與基於高階累積量的時延估計方法作了比較分析。
  2. Experimental results show that the cascading of the speech enhancer and a hidden markov model ( hmm ) based speech recognizer can significantly improve recognition accuracy in noisy environments without performance degradation for clean speech

    通過3種不同的增強演算法用於純凈語音和3種類型帶噪語音的實驗結果分析比較表明，這一方法對純凈語音的識別精度幾乎沒有任何改變而大大提高了系統的抗噪聲性能。
  3. 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。
  4. Experimental results in different noises and snr indicated that this vad algorithm can divide speech segments from non - speech segments accurately and reduce voiced - unvoiced error obviously. ( 2 ) an improved dct - hn speech decomposition algorithm based on the harmonic - noise model is presented

    不同噪聲、信噪比下的實驗結果表明，該演算法可以準確區分語音段與非語音段，明顯降低了基音檢測中清濁誤判現象的發生; （ 2 ）基於「諧波-噪聲」模型提出了一種改進的dct - hn語音分解演算法。
  5. 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。
  6. The saint porch company during the development development home use product, ( the home use product includes ; the down - flow pipe sound insulation, the soundproofing suspended ceiling, the soundproofing wall, the soundproofing floor, the sound - insulated gate, the window and so on ), meanwhile displays the acoustics decoration specialized special skill, meets each acoustics question for the project in, provides the specialized design, the specialized construction, specially aims at the theater, multi - purpose hall, singsong house, pronunciation faculty working office, conference room, recording awning, restaurant, hotel and so on, but also has the equipment level ( between equipment ) the noise quite big place, carries on the specialized design, the construction and the transformation, and achieves the ideal effect

    聖軒公司在開發研製家用產品同時， （家用產品包括；下水管隔聲，隔聲吊頂，隔聲墻體，隔聲地板，隔聲門，窗等） ，同時還發揮聲學裝飾專業特長，為工程中所遇到的各種聲學問題，提供專業設計，專業施工，特別是針對影劇院，多功能廳，歌廳，語音教研室，會議室，錄音棚，酒樓，飯店等，還有設備層（間）噪音比較大的場所，進行專業設計，施工及改造，並且達到理想的效果。
  7. Speech enhancement method based on masking properties of the human auditory system is used to reduce the white noise in the front - end

    摘要為了提高噪聲環境下說話人識別系統的識別性能，將基於聽覺掩蔽效應的語音增強技術作為預處理器，對語音信號首先進行降噪處理，提高輸入信號的信噪比。
  8. As the pitch doubling and halving problems of nccf algorithm often occurred with varied noises and signal to noise ratio ( snr ), vad algorithm is employed to separate speech and non - speech segments

    針對nccf基音檢測演算法在不同噪聲、信噪比下容易發生清濁誤判的問題，本文在基音檢測前端引入語音檢測演算法劃分語音段與非語音段。
  9. 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。
  10. 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。
  11. The traditional detection algorithm, based on zero - crossing or energy, will not acquire ideal effect when the signal - to - noise is low or the signal is weaker. therefore, to resolve the real problem in the real environment that all kinds of random noise and speech signal exit together, some new algorithm must be put forward. account for the complexity of real noise, we integrate the wavelet transform and high - order statistics and advance a new algorithm ; the algorithm can effectively separate the speech signal and the non - gauss noise

    基於過零率和能量的傳統檢測演算法，在噪聲環境比較復雜的情況下效果很不穩定，尤其是信噪比較低或者語音信號較弱時，檢測效果很不理想，因此，在多種語言和噪聲隨機出現、噪聲和語音強弱不一的實際噪聲環境下，必須利用新的演算法提取有用信號和噪聲信號的有效特徵，才能解決實際的問題。
  12. The apsp produced abroad is made by numerical controlled machine tool, which has noise level of 71db ( a ), the apsp produced in our country is made in the method of exploratory which has noise level of 73db ( a ) and 75db ( a ). in order to analyze the influence of stator curve to noise, the author used tri - coordinate measuring instrument to measured exactly the inside surface of stator and got the straddling point coordinate, and made curve fitting by using matlab as language and studied the fitting effect and then worked out the equation of stator transition curve in return seeking, then comparied this method with the standard style and made the conclusion : the equation of atator transiting curve of apsp made by numerical controlled machine tool is close to theorical 5 power curve standard style, but compared with it, the stator transiting curve of apsp produced in exploratory has a major error. combining the testing results of noise, one can know that the qualily of stator transition curve play an importance to t he noise of the pump

    國外生產的汽車動力轉向泵是用數控機床加工的，其噪聲值為71db （ a ） ，國內生產的汽車動力轉向泵是用靠模方法加工的，其噪聲值分別為73db （ a ）和75db （ a ） ，為了分析定子曲線對泵的噪聲的影響，本人用三坐標測量儀對定子內表面進行精密測量，獲得定子內表面的離散點坐標，以matlab語言為工具對離散點進行曲線擬合，觀察擬合效果，然後，用回歸方法求出了定子過渡曲線的方程，並把該方程與理論方程標準型進行了比較，得出如下結論：用數控機床生產的汽車動力轉向泵的定子過渡曲線方程非常接近理論5次曲線標準型，而用靠模方法加工的汽車動力轉向泵的定子過渡曲線與理論5次曲線標準型相比較，則存在著較大的誤差，結合噪聲測試結果可知，定子過渡曲線的優劣，對泵的噪聲大小有著重要的影響。
  13. Melp vocoders utilize mixed pulse and noise as the excitation to elimate the buzzes in traditional lpc vocoders, and add a jitter voicing state to overcome the tonal noise. parameters " interpolation, adaptive spectrum enhancement and pulse dispersion also are adopted to improve the continuity. the synthetic speech of melp vocoders sound much more natural and perceivable than the traditional vocoders "

    Melp聲碼器採用混合脈沖和噪聲激勵解決了經典lpc的嗡嗡聲的問題；引入了抖動濁音狀態以克服音調噪聲；利用參數插值、脈沖散布和自適應譜增強等措施提高合成語音的自然度和可懂度；此外還採用了多帶激勵，使其具有了比較強的抗背景噪聲的性能。
  14. The al - di ( articulation index weighted - directivity index ), which is a measure for the improvement of the signal - to - noise ratio, is increased by about 30 percent for the trimictm technology. this corresponds to an improvement in speech intelligibility of up to 30 %

    研究亦顯示，在吵雜環境中，即語音與噪音的比率較差時，方向性麥克風對聽障者尤其有效，言語理解能力可提升至30 % ，使聽障者更容易與人交談。
  15. ( 3 ). a new method based on masking properties of human ear for speech enhancement is proposed. ( 4 ). the proposed methods for speech enhancement are implemented in computer simulation. and the result is satisfactory

    在上述工作的基礎上，對各語音增強方法，分別在白噪聲和有色噪聲條件下，在- 10db 10db信噪比范圍內進行了計算機模擬實驗，得到了令人滿意的結果。
  16. The matching pursuit techniques are applied to enhance speech signal, and a method to determine the threshold of coherent ratio is provided in the enhancement procedure based on matching pursuit. with the method, the noisy signal can be efficiently enhanced in a rather wide range while the statistical property of signal and noise is unknown

    運用匹配跟蹤技術處理了語音信號增強問題，給出了匹配跟蹤信號增強過程中相干比閾值的確定方法，實現了在未知信號與噪聲統計特性的情況下，在相當大的范圍內明顯增強信號的目的。
  17. We made an improvement in overcoming the defects in speech signal adaptive delta modulation ( abbr. adm ), such as slope overloading and grain noise. in this method, numerical sliding average filtering was used for filtering decoding speech signal. experiments and analyses indicate that the method makes waveforms in good agreement between the decoding of adm and the original pulse coding modulation ( abbr. pcm ) signal, and considerably improves, the playback speech quality in naturalness, legibility and under standability

    針對語音信號自適應增量調制（ adm ）方式中斜率過載和顆粒噪聲缺點，提出了一種改進方法，它利用滑動平均方法對解碼后的信號進行數字濾波.試驗和分析表明，該方法使解碼后的信號波形與原脈沖編碼調制（ pcm ）波形具有很好的一致性，使再生語音質量在自然度、清晰度和可懂度方面比改進前均有較大提高
  18. 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。
  19. The comparison experiments of speech signals corresponding to different snr and noise type was designed using the measure of complexity behaviors based on the information gain

    採用基於信息增益的復雜性行為度量，對含不同噪聲類型，以及不同信噪比的各種中英文語音樣本進行了對比實驗。
  20. After a great amount of detailed computer simulations and concise qualitative and quantitative theoretical analysis, the turbo codes " parameters and fpga specific hardware implementation architecture suitable for being integrated into dtv systems are determined. furthermore, the codec is completely designed with verilog hdl, ending with an occupation of less than a 600 - thousand - gate fpga chip. at this lowest hardware cost, a white noise snr threshold of 1. 8db at a net stream rate of 6mbps is achieved, which exceeds all other existent dtv systems " performance

    經過大量詳細的計算機軟體模擬和簡明扼要的定性與定量的理論分析，最終確定了數字電視系統中適合採用的turbo碼參數及針對fpga特殊構架的硬體實現結構，並用verilog硬體描述語言完成了turbo碼編譯碼器的完整設計，以佔用不到一片60萬門fpga晶元的較少的硬體資源取得了在6mbps凈碼率下1 . 8db的白噪聲信噪比門限這一遠遠超過現有任何數字電視系統的性能。
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