spectrum of speech 中文意思是什麼

spectrum of speech 解釋
言語聲譜
  • spectrum : n. （pl. -tra ）1. 【物理學】譜，光譜；波譜；能譜，質譜。2. 【無線電】射頻頻譜；無線電（信號）頻譜。3. 【心理學】（眼睛的）余像；殘像。4. 〈轉義〉范圍，幅度；（連續的）系列。
  • of : OF ＝Old French 古法語。
  • speech : n. 1. 言語；說話；談話；說話能力（或方式）。2. 民族語言，方言，專門語言；〈罕用語〉流言。3. 演說，演講；發言。4. 【語言學】詞（類）；引語；用語。5. （樂器的）音，音色。
  1. 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
  2. Moreover in speech enhancement, especially in reducing the pulse noise, morphological algorithm has its unique advantage. particularly morphological filter may maintain the preferable accurate of the speech signal in speech waveform, and which produces little impairment to the formant of speech. so the spectrum structure of the speech is retained well, and the quality of the speech will not be reduced

    特別是，在時域波形分析中，形態學濾波增強較小波去噪更好地保持語音信號的細節；在頻域分析中，形態學濾波對語音信號的基音頻率、頻譜斜率、共振峰等語音特徵的影響很小，因而能夠較好的保留語音信號的頻譜結構，使語音品質不致降低。
  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. In order to obtain a high - quality speech codec, the phase information of speech should be included in codec. in this thesis, a method for quantizing the phase of sew ( slowly evolving waveform ) and reconstructing sew ’ s phase with cubic polynomial interpolation is given based on the perceptual weighting analysis - by - synthesis ( a - b - s ) vector quantizer for the phase spectrum in wi coder

    本文基於感覺加權相位譜分析合成（ abs - analysis - by - synthesis ）矢量量化方法，給出了一種wi編碼器中慢漸變波形（ sew - slowlyevolvingwaveform ）的相位信息量化及合成端相位的三次多項式插值重建方法。
  5. 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。
  6. The standard spectrum of chinese speech

    漢語標準頻譜
  7. 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。
  8. The refinements made to the original methodology are based upon not only the abroad new research about noise evaluation, but also upon the thought of the domestic base of current noise evaluation. the procedure involves not only considering the noise sound pressure level, but also considering the speech interference level and noise spectrum balance problems, which offer enough information about building noise environment for engineer and owner

    該評價方法借鑒了國外在噪聲評價方面最新的研究成果，同時也考慮了國內目前的噪聲評價狀況，該方法不僅僅考慮了噪聲聲壓級一個因素，同時還考慮了語言干擾級，以及噪聲頻譜的平衡性等問題，給工程人員以及建築業主提供了建築聲環境方面充足的信息，這是很有意義的。
  9. Secondly, the relativity of the speech signal and that conducted by bone and the relation between the spectrum and timbre were analyzed, so some regulations were discovered. based on correction of spectrum, a method for speech reconstruction was proposed. thirdly, the power constant was obtained by large quantity statistic experiments

    文中分析了骨導信號和語音信號的相關性，以及頻譜與音色的關系，發現了其中的一些規律，提出基於譜修正的語音重構，並通過大量統計實驗，得到骨導信號譜修正的權系數。
  10. Depending on the specific application, the enhancement system may be directed at different objectives. what is contained in the thesis is as following : ( 1 ). a variety of methods based on short - time spectrum estimation for speech enhancement are discussed

    系統地研究了基於語音短時譜估計的各種增強方法，包括幅度譜相減法、功率譜相減法、維納濾波法、最小均方誤差法、兩態軟判決等。
  11. 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。
  12. Spectrum analysis of vehicle noise and speech signal and the application of a digital subsidence filter

    汽車噪聲和語音信號的譜分析及陷波器的應用
  13. 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。
  14. System scheme of speech coding plus spread spectrum communication was presented based on a full analysis of noise characteristic, attenuation characteristic and impedance characteristic of low - voltage power line. spread spectrum carrier ( abbreviated as ssc ) technology is adopted to overcome problems existing in signal transmission over power line. high quality, low rate mbe compression algorithm was used to complete speech encoding and decoding

    在對低壓電力線路的噪聲特性、衰減特性和阻抗特性三個方面充分分析的基礎上，本文提出一種語音編碼+擴頻傳輸的系統總體方案，採用擴頻載波（ spreadspectrumcarrier ，縮寫為ssc ）技術克服電力線傳輸信號存在的問題，採用語音合成質量高並具有較低碼率的mbe壓縮演算法完成語音信號的編解碼。
  15. Such discussion involves a wide spectrum of technique issues, such as functional structures and components, criteria for performance estimation, and the important issues relevant to the ums. chapters 4 and 5 deals with the development of communication software based on the tts and asr techniques with the help of the speech development tool kit, the microsoft speech sdk. in this work, object oriented approach is adopted, and basic modules are encapsulated as activex control components to enable convenient communication among subsystems

    本文第4章重點討論了zhx - ums中與tts asr相關模塊開發問題，較詳細地介紹了模塊功能的實現；第5章進一步討論如何以面向對象的方法為基礎對模塊進行封裝（封裝成activex控制項） ，以解決模塊間通信的問題和與其他子系統間的信息交換問題。
  16. Lsp parameters are quantized by multi - stage vq with fourth - order interframe ma prediction. this scheme has little spectrum distortion, even if the two types of speech have many variations of lsp parameters

    並對涉及到的語音處理的關鍵技術，如線性預測、 lpc與lsp的轉換、矢量量化、基音分析等技術作了深入研究。
  17. We bring forward the method through compare the two projects adaptive spectrum enhancement and wavelet analysis which combines the wavelet analysis and the adaptive theory and have the information of several frequence bands being the reference input signals to reach the adaptive processing and restructure of speech signal by making use of merits of the two theories

    通過對比自適應譜線增強和小波變換的兩種方案，提出了把小波分析理論與自適應理論相結合，利用兩種理論的優勢，用小波分解后的若干層頻段信息作為參考輸入信號，實現語音信號噪聲的自適應處理和重構。
  18. Aiming at the longer delay in the searching minimum of the noisy speech spectrum, a novel minimum band energy approach is proposed to speed up noise update

    針對傳統搜索頻譜最小值方法延遲較大的缺點，提出了最小頻帶能量演算法，加快語音幀內大部分噪聲的更新速度。
  19. The simulation on computer shows that the proposed algorithm is more effective than the searching minimum method of the noisy speech spectrum in variable noise - level environments

    實驗結果表明，該演算法對電平隨時間變化的噪聲取得了較好的增強效果。
  20. In this scheme, the speech residual signal is synthesized directly using magnitude spectrum and phase track where there are continuous changes for pitch between frames, while the speech residual signal is synthesized using phase intergradations with a burst of the pitch

    在該方案中，當幀間的基音周期連續變化時，語音殘差信號由幅度譜和相位軌跡直接合成，而當基音周期發生跳變時，則利用相位過渡過程合成語音殘差信號。
分享友人
分享到 Line

分享到臉書