頻帶濾波器 的英文怎麼說

中文拼音 [bīndài]
頻帶濾波器 英文
bf band filter
  • : Ⅰ形容詞(次數多) frequent Ⅱ副詞(屢次) frequently; repeatedly Ⅲ名詞1 [物理學] (物體每秒鐘振動...
  • : 動詞(除去液體雜質) filter; strain
  • : Ⅰ名詞1 (波浪) wave 2 [物理學] (振動傳播的過程) wave 3 (意外變化) an unexpected turn of even...
  • : 名詞1. (器具) implement; utensil; ware 2. (器官) organ 3. (度量; 才能) capacity; talent 4. (姓氏) a surname
  • 濾波器 : [電子學] electric filter; (electric) wave filter; filter
  1. Electroacoustics. octave - band and fractional - octave - band filters

    電聲學.倍和分倍頻帶濾波器
  2. Electroacoustics - octave - band and fractional - octave - band filters

    電聲學.倍和分倍頻帶濾波器
  3. 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。
  4. A uhf sensor is designed and fabricated, which mostly consists of antenna, uhf amplifier, high pass filter, and broadband wave detector

    通過與大華無線電廠合作,研製了超高傳感,其主要由天線、超高放大、高通、寬等組成。
  5. 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率分析的組、內毛細胞的半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。
  6. An lpf can be simulated by setting the bpf ' s centre frequency to zero

    的中心率設置到0 ,就可以模擬低通
  7. The emphasis of research is decimation technology. design cic filters and halfband filters with this theroy. 4

    重點研究了抽取技術,並將這種技術應用於數字中系統,設計了cic和半; 4
  8. Later on we discuss the basic theory of multi - rate signal processing, and the polyphase algorithm for interplation filter, then produce the efficient algorithms for interplation : half - band filter and cic filter. we analyze their computing quantity and performance, especially the mirror elimination feature and the at - tenuation in the passband

    之後討論了多速率信號處理的基本理論,比較了不同的內插演算法,分析了半和cic內插的計算代價、時性能,以及各自的抑制鏡像特性。
  9. First, how to conduct sample and quantification of continuous time signal which is prior condition of sdr is explored in detail, and the comparison and analysis of some sample modes are given in which band pass signal sampling theorem is most important. second, multi - sample rate signal processing which is an important basis of sdr is studied. emphasis are put on decimation and interpolation those are the most fundamental process and the realization of decimation and interpolation filter

    在基於中采樣的軟體無線電結構框架下,首先詳細探討了軟體無線電的前提條件,即如何對連續時間信號進行采樣量化,比較分析了幾種采樣的方式,其中最為重要的是通信號采樣定理;然後探討了軟體無線電的一個重要基礎,即多采樣率信號處理,重點討論其最基本的兩個過程抽取和內插以及抽取和內插的實現;接著介紹了結構簡單、適用於一級抽取的cic和適用於做2倍抽取的半;再次論文在總結了傳統的調制解調基礎上,結合軟體無線電件的特點,系統的探討並實現了基於正交思想的am 、 fm 、 ask 、 fsk 、 bpsk 、 qpsk的正交調制解調演算法。
  10. Video coding using m - channel filter banks

    組的視編碼方法
  11. Our treatment shows in the raman effect case, mnls solitons, the same as nls solitons, keeps its energy, initial center and initial phase unchanged. however, the raman effect, not the same as nls solitons. reduces its amplitude, widens its width and under the same conditions, the self - frequency shift of mnls soltions is closer to the result of numerical simulation. in the raman effect together with frequency filters case, mnls solitons, the same as nls solitons, keeps initial center and initial phase unchanged and bandwidth - limited frequency filters can make the mean frequency of mnls soltion stand a steady value at the red side of the initial mean frequency, i. e. suppress the self - frequency shift of mnls soltions. and that, the other physical parameters of mnls soltion last stand a steady value. however under the same conditions, the steady value of the mean frequency of mnls soltions is closer to the initial mean frequency and the result of numerical simulation

    有所不同的是, raman效應雖然不改變孤子能量,但會引起孤子峰值的下降和寬度變寬,且在同一條件下, mnls孤子微擾理論得到的自移比nls孤子微擾理論得到的更接近直接數值計算結果。有限寬的也不引起mnls孤子初始中心、初始位相的改變,選取適當的參數值能使孤子的平均率穩定在初始平均率的紅側一穩定值,抑制了自移,而且孤子的各物理量最後都穩定在一穩定值,這些與nls孤子微擾理論都是類似的。有所不同的是,寬度為飛秒量級下, mnls孤子平均率的穩定值更接近初始平均率,更接近數值計算結果。
  12. Chapter 2 introduces the principle theory of lna, harmonic mixer, multiplier, spdt, vco and the basic design flows of the ads examples ( x - band ), the power combine technology, the millimeter - wave power amplifier mmics ’ trends nowadays

    第三章介紹了毫米前端中無源電路的設計,包括毫米的設計、微和中的設計、導到微的過渡、微信號的層間過渡。
  13. Octave bandwidth filter

    程寬
  14. Electroacoustics - octave - band and fractional - octave - band filters ; amendment 1

    電聲學.倍和分倍頻帶濾波器.修改件1
  15. Broad band filter

    頻帶濾波器
  16. Electroacoustics - octave - band and fractional - octave - band filters iec 61260 : 1995 a1 : 2001 ; german version en 61260 : 1995 a1 : 2001

    電聲學.倍和分倍頻帶濾波器
  17. This paper presents a model of cosine basis functions neural network based on bp algorithm, discusses the relation between the algorithm of neural network and amplitude - frequency characteristic about the linear phase fir filter, introduces the convergence condition of neural network algorithm, and studies the optimal design example about the high - order fir double - band - pass filters

    提出了一種基於bp演算法的正弦基函數神經網路模型及演算法的收斂條件,研究了該神經網路演算法與fir線性相位特性的關系,給出了高階雙通的優化設計實例。
  18. In the next, we discuss the system of the meg - 1 layer i. the paper centers on the two kernel sub - parts : filtering coding and psychoacoustic model, do some research work in sub - band coding ( cbc ) theory and the relate theory such as quadrature mirror filter ( qmf ) and analyse sub - band filter ; also do research work in psychoacoustic theory especially the part related to the mpeg - 1 layer i. in the third chapter, introduce the ti tms320c6000 series dsps and their characteristics, also about the software development flow and the ti dsp / bios operating system of it. the forth chapter is the most important, firstly, according the algorithm flow in protocol, using c language validate the algorithm ; then, transplant and optimize the coding in dsp. in the processing of optimize, acording the assembler program characteristic of ti dsp, the paper put forward the analyse sub - band filter dsp optimization algorithm base on the eight spot idct. the algorithm has been optimize have greatly improved the work efficiency. make use of the technology of the dsp / bios host channels, data io pipe, software interrupt, we implement the musicam algorithm base on dsp / bios

    論文首先對當前語音編碼技術的發展、分類以及mpeg系列音標準作了介紹;接著在第二章,給出了layer的musicam ( masking - patternuniversalsubbandintegratedcodingandmultiplexing )演算法的系統組成,圍繞分析子和心理聲學模型兩個核心模塊,深入研究了子編碼工作原理、比特分配及子編碼中用到的正交鏡像和分析子;探討了心理聲學基本原理和mpeg . 1layer所用到的心理聲學模型。第三章對titms320c6000系列dsp作了簡介,介紹了6000系列dsp結構特點、 c6000dsp軟體開發流程和tidsp / bios操作系統。第四章是本文的重點,首先根據協議給出的演算法用標準c語言編程實現並調試通過。
  19. In view of the importance of rf front - end circuits in the receiver system, the paper finally is focused on the analysis and design of the rf front - end circuits, including the design of low noise amplifier ( lna ), microstrip filter and balanced mixer. all the circuits above are simulated individually using ansoft serenade, and the results are satisfying with the desired performance

    考慮到射前端電路性能的好壞會直接影響到整機的性能,文中還重點論述了該接收機射前端電路的分析和設計,主要包括低噪聲放大、微和單平衡混的理論分析和實際設計,並在ansoftserenade環境下進行了模擬模擬,模擬結果符合設計要求。
  20. This approach the problem of conquers phase excursion and frequency drift that anciently adopts fixed frequency filter and its frame is simple

    這種方法克服了以往採用固定來的相移和率漂移問題,而且結構簡單。
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