高音增強 的英文怎麼說
中文拼音 [gāoyīnzēngqiáng]
高音增強
英文
treble boost-
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。Audio localization technology, a new intersecting subject, touches on such research fields as psychoacoustics and physiology, artificial intelligence and high - capability computer system and so on. and it can help transfer and identify visible information, enhance the fidelity, imagination and immersion of 3d simulation environment. so it has a broad application future
音頻定位技術作為一門新興的邊緣交叉學科,涉及聽覺心理學、聽覺生理學、人工智慧和高性能計算機系統等多個研究領域,且具有廣泛的應用前景,它可以幫助傳遞和識別可視信息,增強三維模擬環境的逼真度、想像力和沉浸感,在軍事和民用方面有廣泛的應用。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。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。The dvd recent three stars face toes the chinese consumer, releases the world the section 1 to make to order for high and clear degree television to broadcast the, namely three stars the dvd - hd938, it can pass the general port to realize the pal system the type bottom to is every other line with the line painting quality for scanning outputting 576 is 576 p, can emerging the export surplus commonly even pal system type line scanning ; but the adoption image strengthen the technique to ask for help the dvi port to can more realizes 1080 is 720s p exportation result. more important, pass to see the format conversion of the, and the can of high and clear degree television is produceding can attain of distinguish the rate. 1080 is 720 p
慧聰影音商務網最近三星面向中國消費者,推出了世界第一款為高清晰度電視定製的dvd播放器,即三星dvd - hd938 ,它可以通過一般埠實現pal制式下隔行和逐行掃描輸出576i 576p ,能夠展現出超過普通甚至pal制式逐行掃描的畫質而採用影像增強技術藉助dvi埠更可以實現1080i 720p輸出效果。更重要的是,通過視頻的格式轉換,可以產生高清晰度電視所能達到的解析度1080i 720p 。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。Multi - channel surround sound technology obviously strengthened the music performance and the prominent music essential factors by its own characteristic such as immerse feeling, surrounding feeling, spatial feeling, level feeling, scene feeling, localization feeling, high degree of dissociation and so on, . inevitably, multi - channel surround sound technology is the development tendency of the
多聲道環繞聲以其自身的浸沒感、包圍感、空間感、層次感、現場感、定位感、高分離度等特點,更加突出音樂的要素,明顯增強了音樂的表現力。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。And an automatic gain control ( agc ) loop is introduced to keep the vibrating amplitude of dive tines invariable. it will effectively improve the linearity and robustness of the gyro. moreover, the configuration of the drive circuit is illustrated in detail
3 .設計了微陀螺諧振驅動控制系統,引入了自動增益控制環節,使驅動音叉振動幅值保持恆定,提高了微陀螺的測量線性度,增強了微陀螺的魯棒性,並且詳細介紹了驅動控制電路的實現方法。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。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。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。It solves payment issue through sharing pos and brush card. it solves info sharing and exchanging problem by enterprise application integration it adopts research method of software engineering and uses touch - screen, network, database technology and so on to carry through total design of the system and build the software : it uses user status identify and responsibility control to ensure database and application program ' s security ; it strengthen the code by coding optimize ; it captures and discards application runtime error to enhance the system ' s stability ; it uses multimedia voice and moving picture to show help information, thus makes the system easy to use ; it greatly reduces the maintenance work of the system by self - updating function ; it is an opening system by using star - model network top structure, supporting standard network communication protocol ? tcp / ip and offering standard software interfaced criterion
論文採用軟體工程的研究方法,使用觸摸屏、網路、數據庫等技術,進行了系統總體方案設計和軟體開發:通過對數據庫和應用程序的用戶身份識別和權限控制,保證數據存取和應用程序的安全性;通過對代碼進行優化提高了代碼的健壯性;通過捕捉並拋出系統運行時的異常錯誤提高了系統的穩定性;通過多媒體語音、圖形和動畫提示幫助信息來增強系統的易用性;客戶端程序自動升級功能提高了系統的可維護性,有效地減少了維護工作量;系統採用星型的網路拓撲結構,支持標準的tcp ip網路通訊協議和規范的軟體介面標準,具有良好的開放性。The main research content of the article is involved as follows : ( 1 ) the research and discussion of the quantitative metallographic analysis methods and the measuring methods of micro hardness. ( 2 ) the application of digital image technique in metallographic image preprocess such as gray level transformation, dichotomy, noise eliminating, dilation and erosion, image enhancement, boundary detection, etc. the application of the wavelet and multi - resolution analysis in metallographic image procession to improve the measuring accuracy and efficiency. the application of the region growth and mathematical morphology in analyzing image parameters to improve the flexibility and exaction
本文的主要研究內容: ( 1 )定量金相分析和顯微硬度測量的方法研究; ( 2 )利用數字圖像處理技術,實現金相圖像的灰度轉換、二值化、噪音消除、膨脹收縮、圖像增強、邊緣提取等預處理;引入小波理論、基於數學形態學的區域生長法對采樣圖像進行分析,實現了對採集圖像邊緣的有效提取,從而提高了測量精度; ( 3 )開發了金相圖像分析系統的主體結構(硬體結構和軟體結構) ; ( 4 )採用windows開發平臺的面向對象程序設計語言microsoftvisualc + +進行系統的模塊化設計; ( 5 )提出了採用多模式的知識表示方法建立知識庫,應用正反推理、模糊數學模型、基於規則的模式匹配模型建立金相分析專家系統。In the presence of additive white gaussian noise, an adaptive wavelet - based de - noising algorithm for speech enhancement applications is proposed
針對加性高斯白噪聲的情況,提出一種自適應小波降噪演算法,用於語音信號的增強。The study on the speech emotion recognition has found very important realistic values in such aspects as enhancing the intelligence and humanity of computer, developing new human - machine environment, improving speech recognition results
語音情感識別的研究對于增強計算機的智能化和人性化,開發新型人機環境,以及提高語音識別系統的性能等方面,均有著非常重要的現實意義。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。In addition, we proposed the posterior union model ( pum ), which improves over the conditional union model by retaining the advantage of requiring no identity of the noisy components, and by additionally offering a means of optimally estimating the model order, therefore enhancing the capability of the model for dealing with nonstationary noise
最終本文提出了語音增強結合pum模型的一種新的語音抗噪方法,並且基於這種新方法我們從高識別率和低成本較高識別率兩方面出發,構建了改進型ctranc結合pum新模型和改進型重復wiener濾波結合pum新模型。Speech enhancement experimental results show that this proposed method effectively improves the speech quality and reduces the musical noise
語音增強實驗結果表明,該改進譜減演算法能有效地提高增強效果,更好地抑制音樂噪聲,提高語音質量。分享友人