基于噪聲估計和掩蔽效應的語音增強
發(fā)布時間:2018-01-09 10:37
本文關(guān)鍵詞:基于噪聲估計和掩蔽效應的語音增強 出處:《西南交通大學》2014年碩士論文 論文類型:學位論文
更多相關(guān)文章: 語音增強 噪聲估計 掩蔽效應 諧波恢復 可懂度
【摘要】:數(shù)字化的語音傳送、控制和識別是信息社會的基本組成部分之一。但是語音信號在獲取和傳送途中,都會不可避免的受到各類噪聲的干擾,不僅導致接收者聽到的語音質(zhì)量下降,還會影響語音控制系統(tǒng)和識別系統(tǒng)的正常工作。語音數(shù)字信號處理技術(shù)已廣泛地發(fā)展到了實用階段,語音增強技術(shù)則發(fā)展為該階段需要迫切解決的問題之一。語音增強的目的是消除噪聲干擾和提高語音可懂度。針對不同類型的干擾噪聲,要采用不同的語音增強策略,并且力圖在抑制背景噪聲的同時提高聽者的舒適度。 本文研究是建立在語音增強領(lǐng)域眾多學者的優(yōu)秀研究成果之上的,研究內(nèi)容呈依次遞進的關(guān)系,主要內(nèi)容大致概括如下: 1、簡要闡述了語音增強技術(shù)的基本原理和常用方法,分析了各類噪聲的性質(zhì)和對語音的污染情況。 2、對于平穩(wěn)噪聲干擾情況,本文將二次平滑引入語音活動檢測(VAD)算法中進行后置處理,改善了VAD法估計平穩(wěn)噪聲時出現(xiàn)部分偏差的情況,采用維納濾波來代替譜減法估計純凈語音,避免了“音樂噪聲”的產(chǎn)生。在兼顧了復雜度和處理效果的情況下,該算法可以準確的估計出噪聲并取得較好的增強效果。用多種非平穩(wěn)噪聲對該改進算法進行適用性分析,結(jié)果表明該算法更適用于處理平穩(wěn)噪聲。 3、對于非平穩(wěn)噪聲干擾這一復雜情況,本文研究分析了數(shù)據(jù)遞歸法(DDR),分別用vuvuzela、babble、train和car噪聲對該算法進行仿真試驗,驗證了該算法處理噪聲污染的有效性,同時也證實了本文改進的VAD方法對復雜度和有效性進行了較好的權(quán)衡。發(fā)現(xiàn)了適用于平穩(wěn)噪聲環(huán)境下的增強算法不一定適用于非平穩(wěn)噪聲,但適用于非平穩(wěn)噪聲環(huán)境下的增強算法一定適用于平穩(wěn)噪聲環(huán)境的規(guī)律。DDR算法的有效實現(xiàn)為后文理想二元掩蔽(IBM)算法的研究提供了支持。 4、提高可懂度是語音增強的重要目的。本文研究分析了能夠提高可懂度的IBM算法和諧波恢復(HR)算法。IBM算法是在DDR法估計噪聲方差的基礎(chǔ)上實現(xiàn)的,仿真結(jié)果驗證了該算法提高語音可懂度的有效性。本文采用三級分頻段處理來改進了HR算法改善了傳統(tǒng)HR法卷積運算會產(chǎn)生頻譜混疊的問題。將IBM算法處理后的增強輸出語音作為本文改進HR法的輸入信號進行二次增強處理,有效提高了語音可懂度。
[Abstract]:Digital voice transmission, control and identification is one of the basic components of the information society. But the voice signal transmission way in acquiring and will be influenced by noise inevitably, not only lead to the decline of the recipient to hear the speech quality, also affect the normal work of voice control system and recognition system. The technology has been developed to the practical stage processing of digital speech signal, speech enhancement technology development is one of the urgent problems of the stage. The purpose of speech enhancement is to eliminate noise and improve speech intelligibility. Aiming at the noise of different types, with different speech enhancement strategies, and to enhance the comfort level of the listener in noise suppression at the same time.
This research is based on the excellent research results of many scholars in the field of speech enhancement, and the research contents are progressively progressively related. The main contents are summarized as follows.
1, the basic principles and common methods of speech enhancement are briefly described, and the properties of all kinds of noise and the pollution of speech are analyzed.
2, the stationary noise, this paper will introduce the two smooth voice activity detection (VAD) of the post processing algorithm, part of the deviation appears to improve the VAD method to estimate the stationary noise, using Wiener filter instead of spectral subtraction to estimate the clean speech, to avoid the "music noise" produced in the complex. And the treatment effect of the case, the algorithm can accurately estimate the noise and obtain better effects. Using a variety of non-stationary noise on the algorithm applicability analysis, the results show that the algorithm is more suitable for processing non-stationary noise.
3, for the non-stationary noise of this complex situation, this paper analyzes the data of the recursive method (DDR), respectively vuvuzela, babble, train and car noise simulation test to the algorithm, verify the validity of the algorithm to deal with noise pollution, it also proved that the improved VAD method with a good balance on the complexity and effectiveness are found. The enhancement algorithm may not be suitable for non-stationary noise for stationary noise, but is applicable to non-stationary noise environment and enhance the effective implementation of.DDR algorithm is the Yu Pingwen noise environment is the ideal two yuan masking (IBM) algorithm provides support the study.
4, improve the intelligibility of speech enhancement is an important objective. This paper analyzes can improve the intelligibility of the IBM algorithm for harmonic retrieval (HR) algorithm is.IBM algorithm in DDR estimation method based on the variance of the noise, the simulation results show that the algorithm improve the speech intelligibility is effective. The improved HR algorithm to improve the traditional HR method will produce the convolution spectrum aliasing problem using three frequency processing. This paper will enhance the output speech IBM algorithm after processing the input signal as the improved HR method was two times enhancement, effectively improve the speech intelligibility.
【學位授予單位】:西南交通大學
【學位級別】:碩士
【學位授予年份】:2014
【分類號】:TN912.35
【參考文獻】
相關(guān)期刊論文 前8條
1 牟海維;張芙蓉;;基于聽覺掩蔽效應的多帶譜減語音增強算法[J];大慶石油學院學報;2009年05期
2 呂言國;崔慧娟;;基于改進諧波恢復算法的語音增強方法[J];計算機工程;2012年04期
3 余建潮;張瑞林;;改進增益函數(shù)的MMSE語音增強算法[J];計算機工程與設(shè)計;2010年14期
4 馬建芬;小波變換在語音去噪中的應用[J];太原理工大學學報;2001年03期
5 陳俊,孫洪,董航;基于MMSE先驗信噪比估計的語音增強[J];武漢大學學報(理學版);2005年05期
6 劉曉明,覃勝,劉宗行,江澤佳;語音端點檢測的仿真研究[J];系統(tǒng)仿真學報;2005年08期
7 蔡斌,郭英,李宏偉,龔成;一種改進型MMSE語音增強方法[J];信號處理;2004年01期
8 譚曉衡;許可;秦基偉;;基于聽覺感知特性的語音質(zhì)量客觀評價方法[J];西南交通大學學報;2013年04期
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