濾音器 的英文怎麼說
中文拼音 [lǜyīnqì]
濾音器
英文
acoustic filter-
Nantong sailing marine equipment co. ltd is the professional manufacturer of such equipmentasair - cooler, heatexchanger, evaporator, filter, silencer, airbottle, calorifier, p - ressuretank, boileretc. the products are applied to all kinds of shipping and offshore oil platform
南通賽冷船舶設備有限公司是生產船用熱交換器,壓力水櫃,熱水櫃,消音器,濾器,空氣瓶,鍋爐的專業生產廠家.產品廣泛應用於各類船舶和海上石油平臺The optimal gain filter of ld - celp
低延遲碼激勵語音編碼演算法的最佳增益濾波器The focus is placed on the investigation of the standard of the encoding algorithm for mpeg audio layer iii, and the analysis of the major four modules in the compression algorithm, including encoding of subband filter bank, psychoacoustics model, quantification and huffman coding, frame packing
重點研究了mpeg音頻第層編碼的演算法標準。詳細分析了壓縮演算法中的四個主要功能模塊:子帶濾波器組編碼,心理聲學模型,比特流量化與霍夫曼編碼,幀數據流格式化。In this paper, on the basis of absorption of achievements of the research on auditory physiology, an auditory model simulationg the peripheral auditory system and part of the central auditory system is set up. the model is made of the fitlters presenting the characteristics of the basilar membrane for analyzing the voice signals, the half wave rectification modeling the inner hair cells and energy transfer of nerve fiber
在吸收聽覺生理學研究成果基礎上,建立了一個模擬外圍聽覺系統和部分中樞聖經系統功能的聽覺模型。模型由表徵基底膜的頻率分析的帶通濾波器組、內毛細胞的半波整流特性和神經纖維的能量轉換特性組成,該模型可以作為前端處理來提取語音信號的自相關圖譜。For phonetic signal modulation, if the pass band range of the band pass filter ( bpf ) is 300hz - 3400hz, the anti - noise properties of laser are approximately independent of bias current and parameters of the cavity ; when the pass band range of bpf increases to a certain degree, modulating bias current and parameters of the cavity can improve the anti - noise properties of laser
對語音調制情況,如帶通濾波器的通帶范圍取為300hz - 3400hz ,則激光器的抗噪聲性能基本不依賴于偏置電流和腔內參數;當帶通濾波器的通帶范圍增大到一定程度,調整偏置電流和腔內參數可以實現半導體激光器的高抗噪聲性能。Section ii describes the design approach and implementation of speech module on mcf5249 coldfire core. the speech codec optimizes g. 729a codes and added voice activity detection of g. 729b to save bandwidth ; the implementation of acoustic echo cancellation uses nlms algorithm and it can reduce echo though designing adaptive fir filter and speech detector ; the dtmf and cpt generate signal using two second order digital sinusoidal oscillators and detect signal by picking up the frequency information. but only get the frequency information is not enough in cpt detector, this thesis introduces a method
其中對語音編解碼器的設計採用優化g . 729a代碼達到設計要求,並在此基礎上加入g . 729b的靜音檢測模塊,以進一步降低網路傳輸帶寬;對回聲消除器的設計採用nlms演算法,通過設計自適應fir濾波器和語音檢測器達到回聲消除目的;對雙音多頻設計,信號發生端採用構造靜態參數表並通過二階正弦振蕩器產生信號,信號檢測端提取頻率信息以檢測信號;對呼叫進程音設計,除了類似雙音多頻的信號發生及頻率檢測設計外,還需要檢測信號持續時間,作者設計了一種基於匹配狀態表的方法以檢測信號持續時間。The digital audio processors tas300x series produced by ti have powerful ability to handle the audio. this paper introduces the method of designing the eq filter while introduce the digital audio processors tas300x series. about the control of the system, i2c bus and the msp430 mcu are introduced
德州儀器公司( ti )出品的專用dsp數字音頻處理器tas300x系列,有著強大的音頻處理能力,性價比很高,本文在簡介tas300x系列數字音頻處理器的同時,還介紹了數字均衡濾波器的設計方法。Skj ( jjw ) series of ac purifying regulated power supply, incorporating functions of regulation and auti - inter ference, designed and muters, xerox, video equipments, medical instrmments and other electr onicinstrmmerts and autocontrol systems for is wide vange of regulating, high precicion, quickacting, high effciency, small itcan restrain restrain sharp voltageand protect from moise of electric power net
系列進精密凈化電源是當今國際上普遍採用的現代電源調節技術,配以大容量濾波器與集成化控制系統構成;它具有如下特點:可靠性好、效率高、優良的動(瞬)態,體積小、重量輕、噪音低、負載適應能力強;適用於對電源要求較高的儀器或設備。A feature extraction method based on gauss wavelet filter in speech recognition
基於高斯小波濾波器的語音識別特徵提取方法The echo canceller includes an adaptive finite impulse response filter that generates an adaptive filter weight vector
此?音消除系統由一適應性有限脈沖響應濾波器所組成,它能產生一組適應性濾波器權值向量。Fir ( finite impulse response ) filter is one of the basic algorithms for digital signal processing, which is a kind of important lti discrete - time system, widely used in acoustic processing and image processing area
Fir ( finiteimpulseresponse )濾波器是數字信號處理的基本演算法之一,是一類較為重要的線性時不變系統,廣泛應用於聲音、圖像處理等現代通信技術中。Note : to reduce noise, install an engine intake filter and use a suitable expansion muffler
注:為減少噪聲,請安裝進汽過濾器和適當的消音器。100 high and low pass electronic filters to help tailor the sound to your hearing
100個濾波器根據您的聽力對聲音進行「量體裁衣」 。Analog synthesis often use low - pass filter to remove electronic audio generator generated by high - frequency noise
模擬合成器常常使用低通濾波器來去除電子音頻發生器所產生的高頻噪聲。We select fpga of type xc3s200 as hardware to design the coder and display the hardware resources inside, moreover study the method and steps of designing dsp, based on fpga, by using system generator, finally, it emphasizes the design process of multi - band excitation vocoder. we can work out the module of high pass filter and the module of low pass filter, module of divide frame, module of keynote rough estimate, module of keynote fine estimate, module of band - separated v / u judgment / verdict and module of band - separated amplitude estimate, by using simulink, ise and system generator
本文選用型號為xc3s200的fpga作為設計編碼器的核心硬體,介紹了其內部所含的硬體資源,並研究了利用systemgenerator基於fpga設計dsp的方法和步驟,最後,本文把重點放在多帶激勵語音編碼器的設計上,利用simulink , ise和systemgenerator分別設計其中的高通低通濾波器模塊、分幀疊加模塊、基音粗估模塊、基音精細估計模塊、分帶v / u判決模塊、分帶幅度估計模塊。Dsp digital shift frequency sires is new generation conference system and teaching equipment, which adopts dsp technology. it can lock the point of howl a automatically, dsp digital shift frequency and aptitude mix sound of transmitter
數字自動濾波器,數字話筒智能混音器,數字高低音調節器等多項數字化音頻處理模塊的技術優勢,解決了會議教學擴聲中的聲反饋問題,可提升話筒靈敏度Microfilter is mainly used to separate the data and voice transmission on the same phone line
每一個電話分機已安裝了寬頻濾波器確保資料傳輸不被話音傳輸所影響。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。An optimal design method for the prototype filter design of the filtered multitone modulation system is proposed
摘要給出了濾波多音調制系統的原型濾波器優化設計方法。This method can not only minimize the stopband energy of the prototype filter, but also determines the inter symbol interference power and the transition band energy, which can be used to design prototype filters for both the critically sampled fmt system and the non - critically sampled fmt system
該方法不僅可以最小化原型濾波器的阻帶能量,還界定了符號干擾功率、過渡帶能量,適用於嚴格采樣和非嚴格采樣濾波多音調制系統。分享友人