識別音 的英文怎麼說

中文拼音 [zhìbiéyīn]
識別音 英文
discriminate tone
  • : 識Ⅰ動詞[書面語] (記) remember; commit to memory Ⅱ名詞1. [書面語] (記號) mark; sign 2. (姓氏) a surname
  • : 別動詞[方言] (改變) change (sb. 's opinion)
  • : 名詞1. (聲音) sound 2. (消息) news; tidings 3. [物理學] (音質) tone 4. (姓氏) a surname
  • 識別 : 1 (辯別; 辯認) discriminate; distinguish; discern; tell the difference; spot 2 [計算機] identif...
  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. The display unit which consists of an alphanumeric display and a buzzer, informs the driver of recognition result, timing of uttrance.

    顯示器包括一個數字字母顯示器和一個蜂鳴器,它向駕駛員報告結果和發定時。
  3. The display unit which consists of an alphanumeric display and a buzzer, informs the driver of recognition result, timing of utterance.

    顯示器包括一個數學字母顯示器和一個蜂鳴器,它向駕駛員報告結果和發定時。
  4. Noisy chinese speech recognition based on linear prediction of one - sided autocorrelation sequence

    基於單邊自相關線性預測噪聲中漢語語
  5. In the second part, the focuses are put on the methods of realization of atc instruction speech recognition, automatic pilot reply text - to - speech, management of sky talk dialogue and aircraft state feed back display

    第二部分重點研究了本系統的管制指令語、自動機長語合成、陸空通話管理器和飛行信息反饋顯示四個模塊的具體設計方法及過程。
  6. St andrews university researchers studying in sarasota bay off florida ' s west coast florida discovered bottlenose dolphi used names rather than sound to identify each other

    英國聖安德魯大學的研究人員們在位於佛羅里達西海岸的薩拉索塔灣進行了此項研究,發現寬吻海豚並非通過嗓,而是通過「名字」 (叫聲的內容)來對方。
  7. St andrews university researchers studying in sarasota bay off florida ' s west coast florida discovered bottlenose dolphins used names rather than sound to identify each other

    英國聖安德魯大學的研究人員們在位於佛羅里達西海岸的薩拉索塔灣進行了此項研究,發現寬吻海豚並非通過嗓,而是通過「名字」 (叫聲的內容)來對方。
  8. In this diploma thesis, the statistic and structural characteristic of musical score image is analyzed and synthesized by relevant technology of image project, pattern recognition, mathematical morphology, software engineering, music knowledge, midi and so on. the concept of direction number has been defined, and then the mathematical morphology theory is used to process musical score image and recognize musical information. specialized direction number algorithms are firstly used to preprocess a musical score image and then recognize stafflines, barlines, pitch, note values, clef, etc. finally the musical information of the musical score image is automatically stored in the midi format

    本文利用圖像處理、模式、數學形態學、樂知庫與midi等相關技術,分析與綜合數字樂譜圖像的統計與結構特徵,提出了方向數等概念,對樂譜圖像進行處理,利用直方圖技術與方向數演算法譜線、小節線、符乾等樂譜的主要框架,然後用數學形態學理論識別音高與時值,最後根據這些樂信息,組合成樂樂譜信息,並自動轉化成midi格式。
  9. Enhancing chinese speech recognition for cochlear implant users by using hearing aid in the contralateral ear

    聯合使用助聽器和增強電子耳蝸的使用者的中文語
  10. ( 3 ) study the segmentation and recognition of audio frequency signal. audio signal can be divided into segments based on zero - crossing rate. ( 4 ) a audio recognition arithmetic based on mfcc is proposed

    頻信號的處理作為項目的一部分,根據要求實現了對單一頻信號的,用vc6 . 0來實現。
  11. ( 2 ) study the character of audio signal. analyze the zero - crossing rate and mfcc. a mean mel coefficients is proposed, it can be used to recognized different audio signal

    通過對mfcc系數進行分析,均值mfcc系數作為頻特徵,採用動態時間規整演算法,能夠對單一頻進行,對已有數據源進行測試,有較高的率。
  12. The initial vsd process uses two main characters, the average instantaneous energy and the average instantaneous zero crossing rate ( zcr ). to make the first recognition for the start and the end, the emphasis of which is to select the appropriate value of the threshold and the length of frame. in the final vsd process, the author compares several characters and confirms the new recognition character

    初步分段過程使用了能量和過零率這兩個主要特徵進行端點檢測,重點是合理選擇兩個重要參數? ?門限和統計幀長度的取值;在最終分段過程中,筆者首先通過比較幾種特徵的效果,選擇卡爾曼濾波參數作為再次分段的特徵,還提出了一種新的特徵? ?周期性緩變特徵,使用這兩個特徵分在子語段內進行端點檢測。
  13. The speech recognition ' s system ( in this paper we mainly discuss ibm viavoice ) has the certain capacity of self - adaptation to the speech velocity, volume and tone, but the capacity of those is not enough with different enunciator

    目前的語系統(本文中主要是指ibm的viavoice語系統)對語速、量和調都具有一定的自適應調整能力。
  14. 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「模糊譜相減法語增強」 ;針對語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、利用共振峰參數進行第二級的兩級漢語數碼語系統,在保證系統實時性的同時,實現連接漢語數碼語系統率的提高;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。
  15. Firstly, we study the construction of emotion - speech template database, and analyze the common features such as pitch, energy and formant. after choosing the useful features by using fuzzy entropy effectiveness analysis, we get better performance with the application of neural network. in addition, we propose some more efficient features such as speech rate, pitch slope, mel - frequency cepstral coefficients and its transient parameters, and design a processing model based on vector quantization for cepstral features to fusing different features

    本文首先介紹了情感語數據庫的建立情況,然後研究了基頻率、振幅能量和共振峰等目前常用的情感特徵在語情感中的作用,並且通過一種基於模糊熵的特徵有效性分析方法進行了有效特徵的篩選,應用人工神經網路建立了初步的語情感模型,經過實驗發現特徵篩選后系統的效果有著一定程度的提高。
  16. At the same time according to the low recognition rate in speech recognition system, the author used the method of fundamental frequency analysis to build male / female recognition model respectively

    同時針對語系統中率不高的問題,採用基頻率分析的方法分建造男女聲模型。
  17. The topological structure is introduced to analyze homograph qualitatively, the algorithm is robust and insensitive to noises of images. the geometrical structure is used to analyze homograph quantitatively, the fine discrimination between planar objects can be shown

    方法對類似形應用拓撲結構進行定性分析,對噪不敏感;同時結合幾何結構對類似形進行定量分析,能反映平面立體形狀的細微差
  18. Inhibitory processes in the recognition of homophone meanings in chinese

    漢語同異形詞意義中的抑制過程
  19. Speech recognition is a cross - disciplinary, is gradually becoming it, the key to human interface technology, voice recognition and voice synthesis technologies combining technology integral voice response systems, voice synthetic systems, interactive toys, enable people to voice ordering through operation, the control mode flexibility, a broad market prospect

    是一門交叉學科,正逐步成為信息技術中人機介面的關鍵技術,語技術與語合成技術相結合組成語應答系統、語合成系統、互動式玩具等,使人們能夠通過語命令進行操作,控制方式靈活方便,具有廣闊的市場前景。
  20. An error - tolerant algorithm in decoding module of mandarin continuous speech recognition is examined to correct substitution, insertion and deletion errors in acoustic recognition

    摘要本文研究了漢語連續語識別音字轉換中的容錯演算法,以糾正聲學的替代、插入、刪除錯誤。
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