語音演算法 的英文怎麼說

中文拼音 [yǔyīnyǎnsuànfǎ]
語音演算法 英文
phonetic algorithm
  • 語 : 語動詞[書面語] （告訴） tell; inform
  • 音 : 名詞1. （聲音） sound 2. （消息） news; tidings 3. [物理學] （音質） tone 4. （姓氏） a surname
  • 演 : 動詞1 （演變; 演化） develop; evolve 2 （發揮） deduce; elaborate 3 （依照程式練習或計算） drill;...
  • 算 : Ⅰ動詞1 （計算數目） calculate; reckon; compute; figure 2 （計算進去） include; count 3 （謀劃;計...
  • 法 : Ⅰ名詞1 （由國家制定或認可的行為規則的總稱） law 2 （方法; 方式） way; method; mode; means 3 （標...
  • 語音 : speech sounds; pronunciation; voice
  1. 3 ) we try to import the bayesian adaptation, which is widely used in speech recognition, into speaker verification. we use bayesian maximum a posteriori estimation training a speaker model from background model, to solve the problem of model miss matching in speaker verification system

    3 ）為了解決說話人確認中存在的模型不匹配問題，嘗試將語音識別中的貝葉斯自適應演算法引入到基於高斯混合統一背景模型的說話人確認系統。
  2. An agonic algorithm of speech logarithm spectrum envelope

    一種語音信號對數幅度譜包絡的無偏演算法
  3. On the basis of the study on the speech coder algorithms, paper describe an advanced method of developing dsp system software, and as the guidlines, we developed the programme of whole decoder unit. paper stress on analysis of the ecu in decoder unit. aiming at amr algorithms disadvantage of angularity of synthetical speech, paper study on the specutral extrapolation which apply to extrapolate reflect coefficient of track model to make error conceal processing of amr. at last paper analyze existing echo cancellation algorithms using on mobile communication system

    在此基礎上，描述了一種較為先進的大型dsp系統程序開發策略，並以此為指導思想，以美國ti公司c6000dsp開發平臺開發出了整個amr解碼器單元的系統程序。論文對amr解碼器的誤碼隱藏處理單元進行了重點分析，針對原有演算法合成語音自然度不好的缺點，論文研究了將譜外推法應用到amr演算法中外推出聲道模型反射系數參數進行誤碼消除處理。
  4. The hardware of the ip phone codec to be designed is based on the fixed point digital signal processor ( ti ' s tms320vc5410 ) while the compression and decompression core in the software of dsp is based on the itu - t vg. 729a. ip phone codec carryout the task of collecting / playing - back. coding / decoding of speech signal and communication with embedded cpu. etc

    該語音編解碼器的硬體基於tms320vc5410 ，編解碼演算法遵循itu - tg . 729a協議，能夠實現語音信號的採集/回放、編碼/解碼以及同嵌入式cpu通信等功能，在8kbit / s的碼率下能夠提供獲得良好的語音質量。
  5. The optimal gain filter of ld - celp

    低延遲碼激勵語音編碼演算法的最佳增益濾波器
  6. The reference dictionary is sorted phonetically, making a systematic collation algorithm that will follow the same order impossible

    參考詞典是按語音順序排列的，使得系統的排列演算法無法按照相同的順序實現。
  7. 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。
  8. After about two years " insisting and hard working, this goal set at the beginning has become true. the developed c54x general assembly program for g. 729 speech signal compressing algorithm has passed the tracking with more than 3, 000 unitary standard measuring vectors. g. 729 speech signal compressing compiler using c54x general assembly program has been accomplished real - timely, and undistorted rebuilt speech signals have been obtained

    因此本課題選用c54x的通用匯編語言編程實現g . 729語音壓縮編碼演算法，調試並通過了統一標準測試矢量三千多幀，最終在5402開發實驗板上實時實現了g . 729語音壓縮編碼器，獲得未失真的重建語音信號。
  9. And a new pitch extraction algorithm, an active / inactive frame decision algorithm and a voiced / unvoiced frame decision algorithm are developed with the aims to improve the quality of the vocoder and reduce its overall computation load

    本文引入了一種新的基音周期計算方法，靜音幀判決演算法，清音幀判決演算法，清濁音信息接收端重建等新演算法，提高了合成語音的質量，降低了演算法的總計算量。
  10. 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。
  11. In this thesis, the research focuses on pitch detection techniques of the low - rate wi speech coding. aimed at the problems of voiced - unvoiced error, pitch doubling and halving, accuracy of pitch detection and pitch quantization, a series of pitch detection techniques including pre - processing, pitch detection and pitch quantization were proposed

    本文就低速率wi語音編碼中的基音檢測技術進行了深入研究，針對基音檢測中的清濁誤判、基音加倍減半、基音檢測精度及基音量化問題，提出了包括基音檢測前端處理、基音檢測演算法及基音量化的一整套基音檢測技術。
  12. Annex b introduce a voice activity decision ( vad ) algorithm which class speech signal as voice signal and background noise signal

    Annexb提出了一種靜音壓縮演算法（ vad ） ，它將語音信號分為話音信號和背景噪聲信號。
  13. Based on all these above, the results of main research are as follows : ( 1 ) a voice activity detection ( vad ) algorithm based on dct band - partitioning spectral entropy is proposed

    本文以此為基礎，並取得了如下研究成果: （ 1 ）提出了基於dct分帶譜熵的語音檢測演算法。
  14. It synthesizes the excellence of wave coding and parameter coding, adopts vector quantity, analyse - synthesize, perceptual weighting, therefore, gains good speech coding quality at 8kbit / s. cs - acelp can be used in individual telecom, iphone, c / n, microwave telecom and isdn

    Cs - acelp演算法綜合了波形編碼和參數編碼的優點，以自適應預測編碼技術為基礎，採用了矢量量化、合成分析和感覺加權等技術，在8kbit / s速率上獲得了較高的語音編碼質量。
  15. It is expected to be used for 3g personal handy - phone system as standard algorithms which encode speech signals and decode it. additionally, this kind of algorithms which own excellent quality can be application in viewphone and video order programme etc. the thesis introducethe algorithm structure of g. 729

    該協議在可預見的將來可能應用於三代移動通信系統中作為語音編解碼演算法。另外，由於其良好的性能也可應用在多媒體系統中如：可視電話，視頻點播等。本論文概要介紹了g . 729協議的演算法結構。
  16. Voice activity detection method based on gray correlation analysis algorithm

    一種基於灰關聯分析法的語音激活檢測演算法
  17. Theoretical expatiate on general concepts and fundamental principles of information hiding and steganography, also point out possible directions for further research, also analysis the probability of speech as the host carry signal and efficient masking characteristics of psycho - acoustic model, it is shown that : there is an improvement on imperceptibility according to human auditory masking effect

    闡述了信息隱藏技術和隱寫技術的重要概念、基本理論以及廣闊的應用前景。分析了將語音信號作為宿主載體信號的可行性，參考心理聲學模型的特性，得出結論:基於人耳聽覺掩蔽效應的隱寫演算法，在隱蔽性上有很大的提高。
  18. The thesis makes researches on technologies of call centers, voice disposing technologies and the synthetic application of these technologies. with setting up a call center which is called epcc ( electric power call center ) in a electric power company, the thesis describes pivotal technologies in call centers, such as voice disposing technologies >, crm ( customer relationship management ) and web technologies, the thesis represents standard schemes and standard frames of call centers, and the thesis describes pivotal technologies in voice disposing procedures, such as speech synthesize and speech recognize, and the thesis describes voice disposing technology " s applications in call centers that are called ivr ( interactive voice response ) systems, then the thesis discusses the acd ( auto call distribution ) program in epcc

    本論文通過建立呼叫中心的一個實例（電力呼叫中心） ，對呼叫中心、語音技術及其綜合應用進行了較為深入的研究。通過呼叫中心的電力應用，較詳細地論述了呼叫中心的關鍵技術：語音技術、客戶關系管理crm和web技術等；較深入地闡述了呼叫中心的典型方案和典型結構；較詳細地論述了語音信號處理技術的關鍵技術：語音合成技術及語音識別技術：深入討論了語音技術在呼叫中心的具體應用- - - ivr系統及其關鍵技術；較詳細地討論了電力呼叫中心中所採用的acd演算法，並基於acd演算法完善了呼叫中心的表現形式。
  19. A study on prosodic boundaries location and synthesized units selection algorithms in mandarin speech synthesis

    漢語韻律邊界定位與選音演算法研究
  20. It can satisfy the voip application demand at enterprise level and then performs an adjustment test according the vital factors bringing big affection to the quality of voice, such as voice quality and delay of different encoding styles, packaging capacity and delay, echo cacellation algorithm and digital voice mixture algorithm

    此網關能夠滿足企業級的volp應用要求。並在此網關設備上針對對語音質量影響比較大的關鍵性因素進行了調整測試，如各類編碼方式的話音質量與延遲、組包大小與延遲、回聲消除演算法、數字混音演算法等。
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