語音音高 的英文怎麼說
中文拼音 [yǔyīnyīngāo]
語音音高
英文
pitch of eech sound-
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，嘗試將語音識別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。Focused on the application of the generalized cross correlation ( gcc ) time delay estimation ( tde ) in the car microphone array denoising system, gcc tde algorithm is analyzed theoretically and compared with high rank cumulation tde
摘要針對汽車環境中麥克風陣列語音去噪系統的應用，對廣義相關時延估計方法進行了模擬，並與基於高階累積量的時延估計方法作了比較分析。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。In various speech character parameters, formant frequency, bandwidth and pitch frequency are chosen as voice character parameters. the reasons are as follows : hearing apperceive experiments indicates that formant frequency can stand for a majority of voice information, while average pitch frequency can explain 55 % ability of speaker verification
數據結果與多項式回歸和線性多變量回歸相比，支持向量回歸既提高了泛化性能又避免了頻譜不連續性，從而使轉換語音與目標語音的頻譜距離失真分別減少了33 . 29 %和35 . 24 % 。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音數據庫的建立情況，然後研究了基音頻率、振幅能量和共振峰等目前常用的情感特徵在語音情感識別中的作用，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音情感識別模型，經過實驗發現特徵篩選后系統的識別效果有著一定程度的提高。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
同時針對語音識別系統中識別率不高的問題，採用基音頻率分析的方法分別建造男女聲識別模型。Furthermore, the network technique only support the voice service cannot meet the communication requirements of people. people hope to obtain multimedia service such as packet data, video and picture phone etc besides voice service. all these need operators and researchers to look after the mobile communication scheme which optimizes spectrum efficiency and expands system capacity to accommodate more users in major metropolitan markets
此外，僅支持語音業務的網路技術已不能滿足人們對信息交流的需求，人們希望能隨時隨地獲取除語音之外的數據、視頻和圖像等多媒體業務，這些因素都促使運營商和研究者尋求頻譜利用率更高、通信容量更大的移動通信解決方案。With the developing of vlsi in recent years, high function dsp has been produced ( such as tms320 series dsp produced by ti ) and their cost is dropping. thus, this established the foundation for making complex speech coder practical and producible. the paper researched and discussed the fix - point real implementation of g. 728 by dsp tms320c5402 chip
但是，近幾年來，隨著大規模集成電路（ vlsi ）的發展，已生產出高性能數字信號處理晶元（例如ti的tms320系列dsp晶元） ，而且其成本在不斷降低，這就為復雜的語音編碼器的實用化和產品化奠定了基礎。European queen size bed first class shower facility deep soaking tub telephones, idd with dual lines, voice mail system, facsimile machine, high speed internet wireless access, electrical outlets with internet data port for connection of laptop computer, comfortable work desk, personal safe, complimentary coffee tea - making facility and refreshment center. magic kitchen facility with european style cookware
北歐風情皇后床一流的衛浴設施加深浴缸3部電話機， 2條idd電話線，語音留言信箱，傳真機，無線高速寬頻網， 2個電源插口和手提電腦上網埠，舒適的辦公桌，私人保險箱，免費的咖啡茶和電熱壺，迷你酒吧，更有魔幻廚房可提供全套歐洲進口餐具。Hotel kunlun s guestrooms and suites feature the following amenities and services : high - speed broadband internet access international domestic direct dial fold - out make - up shaving mirror coffee tea - making facilities electronic safe mini bar satellite cable television voice mail bathrobe slippers hairdryer wake - up calls room service bed - side control panel in - room temperature controls shoeshine service
昆侖客房和套間提供下列物品及服務：高速寬頻上網國內外直播電話折疊式化裝鏡電子保險箱小酒吧有線衛星電視語音信箱浴衣拖鞋吹風機叫醒服務房間送餐服務床頭控制面板溫度調節擦鞋服務It reduces the “ music noise ” using the human auditory characteristics, and enhances the hearing quality using the speech spectrum distribution characteristics in the time - frequency dimension
主要以譜減法為基礎，結合人耳的聽覺特性從而減少殘留「音樂噪聲」的影響;結合語音的語譜在時-頻域分佈特性從而提高增強后語音的聽覺質量。Experimental results show that the cascading of the speech enhancer and a hidden markov model ( hmm ) based speech recognizer can significantly improve recognition accuracy in noisy environments without performance degradation for clean speech
通過3種不同的增強演算法用於純凈語音和3種類型帶噪語音的實驗結果分析比較表明，這一方法對純凈語音的識別精度幾乎沒有任何改變而大大提高了系統的抗噪聲性能。A framework for dialectal chinese speech recognition is proposed and studied, in which a relatively small dialectal chinese or in other words chinese influenced by the native dialect speech corpus and dialect - related knowledge are adopted to transform a standard chinese or putonghua, abbreviated as pth speech recognizer into a dialectal chinese speech recognizer
但在實際中，多數人所說的普通話因受其方言背景的影響而不十分標準，這大大影響了語音識別的性能。一種解決方案是，對每種方言都收集足夠多的語音數據然後構造相應的識別器，但由於漢語方言種類多且差異大，時間和成本都是很高的。And a new pitch extraction algorithm, an active / inactive frame decision algorithm and a voiced / unvoiced frame decision algorithm are developed with the aims to improve the quality of the vocoder and reduce its overall computation load
本文引入了一種新的基音周期計算方法，靜音幀判決演算法，清音幀判決演算法，清濁音信息接收端重建等新演算法，提高了合成語音的質量，降低了演算法的總計算量。That is, using a soft thresholding to remove noise components from the wavelet coefficients of the voiced and unvoiced speech in noisy speech respectively in a different way, which is not only removing noise but also is preventing the quality degradation of the unvoiced sounds and enhancing the signal - noise ratio
該方法採用軟限幅函數對濁音和清音信號的小波變換系數作不同的閾值處理，既抑制了噪聲，又減少了語音段信息的損失，提高了信噪比。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。No sooner had the adjutant said this than the old whiskered officer, with happy face and sparkling eyes, brandished his sabre in the air shouting vive lempereur
副官語音一落，這位鬍髭濃密的老軍官喜形於色，兩眼發亮，高舉軍刀，大呼「萬歲！ 」Based on the slogan of “ science originates from human being ”, waygoing has attracted a large amount of it talents with explorative spirits and practical experiences and advocated modern enterprise idea of “ sincerity, practice, high - efficiency and creativity ” and regarded high - quality products and perfect as the root of waygoing and satisfied the needs of ngn and voip application of telecom carriers and enterprises, which have long been and will be pursued by waygoing forever
科技以人為本，偉功匯集了一批富有開拓精神和時間經驗的業界精英，倡導「誠信、務實、高效、創新」的現代企業理念，視高質量的產品和完善的服務為企業立足之本，以滿足電信業及企事業單位對ngn及ip語音應用的需求為己任這是偉功永恆的追求。A change in stress, pitch, loudness, or tone of the voice ; an inflection of the voice
變音在語調、音高、音量或音調上的變化；聲音的一種轉調A method of pitch mark determination for a speech, includes : acquiring a fundamental frequency point and fundamental frequency passband signals by using an adaptable filter ; detecting a number of passing zero positions of the fundamental frequency passband signals ; and generating at least a set of pitch marks from a number of passing zero positions
一種決定語音音高標記的方法，系藉以找出一語音之一組音高標記，此決定語音音高標記的方法系利用一可適性濾波器取得一基頻點與一基頻帶通訊號；求取基頻帶通訊號之復數個過零點位置；然後經由復數個過零點位置產生至少一組音高標記。分享友人