語音波形 的英文怎麼說
中文拼音 [yǔyīnbōxíng]
語音波形
英文
speech waveform-
In this paper, combined with currently voice coding technique, espically with the fabulosity development of the mixed voice coding and the increasingly utility of the digital signals processor. we investigated the voice coding technique and discussed emphasizedly the technology of variable rate voice coding technology
本文結合當前語音編碼技術尤其是混合編碼技術的驚人發展及數字信號處理器的日益實用化，研究了語音編碼技術，並重點討論了變速率語音編碼技術。論文簡要介紹語音編碼技術中的波形編碼和聲碼器的主要性能。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。According to difference of speech waveform between transition segment and non - transition segment, dividing between sonant is researched using waveform cross correlation. a method of syllable segmentation is presented based on waveform cross - correlation. ( 2 ) pitch detection of speech signals
為解決濁音之間的分割問題，本文根據語音中過渡段與非過渡段語音波形的差異，利用波形互相關性進行了研究，提出了基於波形互相關性的音節分割方法，並進行了實例分析。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。On real - time speech blind signal separation based on time delay estimation and beamforming
基於時延估計波束形成預處理的實時語音盲信號分離研究Analyzing the waveforms and evaluating the practical effect of the system
5 .對調試輸出波形進行分析，並對實際輸出的語音效果進行評估。The quantized lp coefficients are replaced by the unquantized lp coefficients in the frequency domain expression of the feel weighted filter. the error signal has more similar envelope shape, and the hearing effect is better than before because the unquantized lp coefficients have more accuracy than quantized lp coefficients
由於未量化的線性預測系數具有更高的精度，因此，誤差信號通過修正後的感覺加權濾波器以後，具有與語音信號譜更加相似的包絡形狀，從而更好地利用共振峰對誤差的掩蔽效應，達到更佳的主觀聽覺效果。In the phase of training, it gets the sampling data from the wave files which were stored in the voice library by using the mci functions. then calculates the character vector ( 12 ranks of lpc and lpcc ) and trains them by clustering method, so we get the templates used by speech - recognition, this templates were stored in the template library. in the state of recognition, after calculating the character vector of input voice, we compare it with the character vectors of templates, and then find the best one or refuse it
系統的組成模塊與語音識別系統的基本構成模型基本一致，在訓練過程中，通過調用mci （ mcimultimediacontrolinterface ）提供的函數從語音庫中的波形文件中讀取采樣數據，分幀計算出由12維線性預測系數和12維線性預測倒譜系數構成的特徵矢量，並按照聚類的方法進行訓練，得到后續語音識別時需要的模板，存放于模板庫中。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Experiment results also show that our improved algorithm achieves the same perceptual quality as g. 726 standard while our improved algorithm uses a lower bit rate. therefore, our proposal may lead to telecommunication bandwidth saving and storage requirement reduction
經實驗結果驗證，本語音編碼演算法與g . 726語音波形編碼標準相比，比特率下降了15 . 19 %以上，同時兩者的語音質量完全沒有差別。We first introduce the basic methods of speech processing in time domain. emphatically, we describe linear prediction and tonality detection of speech signal. moreover, we discuss the g. 726 speech waveform coding standard in details
本文首先介紹了語音波形時域分析處理的基本方法，對語音波形線性預測和音調檢測技術作了重點描述，著重研究了g . 726語音波形編碼演算法，並在此基礎上，對該演算法進行了某些探討改進，並用vc + +編程，在pc機平臺上予以實現。In this paper, we investigate speech waveform coding technology with emphasis on the g. 726 recommendation of itu - t. based on g. 726, we present a new algorithm. compared with g. 726, our proposed algorithm achieves the same perceptual quality with lower bit rate
本文結合當前商用市場對語音編碼的需求，研究了語音波形編碼技術，重點研究了itu - tg . 726建議，並在此基礎上探討了進一步降低比特率的演算法，使本語音編碼演算法的音質和g . 726演算法的完全一樣，同時，採用本文演算法的比特率低於採用g . 726語音編碼演算法的比特率。Waveform interpolation speech coding is one of the most potential low - rate speech coding algorithms in recent years. with high performance, wi technique has been widely concerned
波形內插（ waveforminterpolation ， wi ）語音編碼是近年來發展起來的一種非常有潛力的低速率語音編碼演算法，因其良好的性能，受到了研究人員的廣泛關注。Waveform interpolation as a great potential speech coder has got much attention
波形內插（ wi ? ? waveforminterpolation ）作為一種極具潛力的語音編碼方法受到了人們的關注。[ 3 ] the vowel ' s self - correlation function has periodicity and the period of this function is the fundamental sound period of this speech. according to this, we propose a method of adjusting the tone in temporal field by adding or deleting the sampling points in the waveform with the whole speech waveform unchanged
（ 3 ）根據母音自關函數具有周期性並且其周期就是語音基音周期這一特性，提出在保持語音波形總體不變的前提下，在語音時域上對語音進行插值或刪值的方法來調整音調。In order to deal with this problem, this paper introduces the author ' s research on some techniques related to speech processing, mainly including three aspects as follows : [ 1 ] in chinese pronunciation, each syllable contains the vowel, the vowel ' s length is the main part in the syllable but the vowel does n ' t contain the important information. according to these characteristics, we propose a method of adjusting the speech velocity by using similar waveform that is found by correlative coefficient in vowel part to lengthen or reduce the vowel part
本文主要介紹了作者針對這一問題所作的關于語音調整的技術與方法的研究工作，其中包括（ 1 ）根據漢語語音發音時每一個音節都含有母音，母音長度占音節長度的主要部分但是卻不包含發音的主要信息這些特點，提出在語音的母音部分利用相關系數尋找相似波形，然後對母音部分進行幾個相似波形的壓縮或擴展的方法來改變母音的長度進而調整語速。A serial generalized morphological filter with multi - structural element is used suppression white gaussian noise or pulse noise embedded in the speech signal. the paper compares morphological speech enhancement algorithm with classical approach on the feature of speech in the frequency domain and time domain
本文針對形態學在數字語音信號增強中的應用演算法研究，採用多結構元素的廣義形態濾波器，主要用於對被高斯白噪聲或正負脈沖噪聲污染的語音信號的濾波增強，深入研究形態學濾波的語音增強演算法在語音時域、頻域對語音特徵參數的影響。The g. 726 speech coding algorithm involves adaptive linear prediction to eliminate the correlation of signal waveform, so as to reduce the resulting bit rate
G . 726建議採用自適應預測編碼，以消除波形信號的相關性，從而降低語音信號表達的比特率。The experments for ct data curves and speech wave curves show that the method proposed in this dissertation is simple and efficient for curves simulation
對ct數據曲線和語音波形曲線擬合的實驗結果表明，這是一種對曲線進行分形擬合的好方法。At last, the experiment uses a signal source as the input of voice, achieves the transmission, gets good results
最後，用信號源模擬語音波形，實現系統的傳輸，並得到了預期的結果，質量良好。分享友人