語音增強 的英文怎麼說

中文拼音 [yǔyīnzēngqiáng]
語音增強 英文
pectorophony
  • 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
  • 強 : 強形容詞（強硬不屈;固執） stubborn; unyielding
  • 語音 : speech sounds; pronunciation; voice
  1. Speech enhancement based on discrete cosine transform

    基於離散餘弦變換的語音增強
  2. A novel speaker normalization method based on formant recovery and mellin transform

    基於子波變換的自適應濾波語音增強方法
  3. 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。
  4. According to the characteristics of the human pronunciation and the speech spectrum distribution in the time - frequency dimension, the paper finds out that there is a shortcoming of the speech enhancement system which is based on the masking properties of human auditory

    根據人的發音特點，通過分析語音的語譜在時-頻域的分佈，發現把聽覺掩蔽效應應用於語音增強時存在不足之處。
  5. An improved spectrum subtraction speech enhancement method

    一種改進的譜相減語音增強方法
  6. In order to reduce the musical residual noise and the background noise, a speech enhancement method based on masking properties of the human auditory system is described. this method uses bark wavelet packet transform to simulate the frequency feature of human auditory model to get the threshold

    本文以最大限度減少殘留噪聲和背景噪聲為目的，採用bark子波分析的方法模擬人耳基底膜的頻率分析特性來進行語音增強，重點進行模擬人耳聽覺掩蔽效應來確定除噪閾值的研究。
  7. Speech enhancement as the front - end processing module is used to improve the signal - to - noise ratio ( snr ) of the input signal for recognition in the latter stages

    為了讓語音識別系統在安靜的環境和有噪聲的環境中都獲得令人滿意的工作性能，研究了一個將語音增強器和語音識別器級連起來的系統。
  8. 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。
  9. A method for gray - level threshold selection based on maximum entropy

    基於閾值的小波域語音增強新演算法
  10. Clarifying and modeling this human ability are the task of computational auditory scene analysis and the progress will guide the applications such as blind source separation and speech enhancement

    對人的這種能力的研究及其功能建模，是計算聽覺場景分析的任務；它對解決盲源分離及語音增強等工程問題有指導意義。
  11. This dissertation is different from traditional speech enhancement methods which are based on noise characteristic such as adaptive noise cancellation or spectral subtraction processing. in this dissertation the speech signal conducted by bone was taken as the object to be studied and the exploitive study on the acoustical characteristic of speech signal conducted by bone was performed by the method of theory combined with experiment. then a proposition about speech reconstruction based on speech signal conducted by bone was presented, and the design of software and hardware was completed

    本文與傳統的基於噪聲特性的自適應噪聲抵消法、頻譜減法等語音增強降噪技術不同，是以骨導語言為研究對象，採用理論與實驗相結合的方法對骨導信號的聲學特性進行了探索性研究，進而提出了基於骨導信號的語音重構技術，並完成了相應的軟硬體開發。
  12. 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。
  13. 3. a variety of speech enchancement algorithms are discussed including wiena filter, spectral subtraction, mmse algorithm and the masking model combined with the spectral subtraction algorithm

    3 .系統地研究了多種語音增強演算法，包括基於短時譜分析的維納濾波法、譜減法和mmse演算法，並研究了基於人耳聽覺掩蔽效應的語音增強演算法。
  14. Helium speech enhancement based on linear predictive coding

    基於線性預測模型的氦語音增強演算法研究
  15. 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。
  16. Based on those studies, a prototype of noise in speech reducing device for wireless communication is designed and implemented, with its effects evaluated

    因此，研究無線語音降噪技術與語音增強演算法、研製無線語音降噪設備對于提高無線語音通信質量有著重要意義。
  17. In signal space, speech enhancement is adopted to effectively suppress the noise and increase the discriminative information embedded in noisy speech signal. however, the speech distortion introduced by enhancement, as well as the residual noise, is a very adverse factor for recognition

    在信號空間，利用語音增強有效抑制噪聲，提高輸入信號中的鑒別信息，但增強帶來的語音失真和增強后的剩餘噪聲是對語音識別非常不利的因素。
  18. ( 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。
  19. Therefore, speech enhancement technique plays an important roll in speech signal processing

    為了改善語音處理的質量，語音增強就成為語音信號處理中很重要的組成部分。
  20. The research content of the thesis is the speech enhancement technique that is used in the acoustic feedback suppresser. firstly, we searched and compared the methods of noise estimation based on vad and updating the noise spectrum continuously, combined them together to make some improvement. secondly, we research on some speech enhance techniques including short time spectrum analysis speech enhance technique and its improvement form, simulated the algorithms and compared them each other

    本論文研究語音增強技術在聲反饋抑制器中的應用，論文的主要工作包括: 1 .對基於vad （ voiceactivitydetection ）的噪聲估計方法和連續更新噪聲譜的方法進行研究和比較，針對模擬結果分析兩種噪聲估計的性能，並將兩者結合起來，做出改進，用於實際的語音增強系統中。
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