頻率編碼信號 的英文怎麼說

中文拼音 [bīnlǜbiānmǎxìnháo]
頻率編碼信號 英文
fes frequency encoding signal
  • 頻 : 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...
  • 率 : 率名詞（比值） rate; ratio; proportion
  • 編 : Ⅰ動詞1 （編織） weave; plait; braid 2 （組織; 排列） make a list; arrange in a list; organize; gr...
  • 碼 : Ⅰ名詞（表示數目的符號或用具） a sign or object indicating number; code Ⅱ量詞1 （指一件事或一類的...
  • 號 : 號Ⅰ名1 （名稱） name 2 （別號; 字） assumed name; alternative name3 （商店） business house 4 （...
  • 頻率 : frequency; rate
  • 編碼 : encoded; code; coded; encrypt; codogram; coding編碼表 encode table; 編碼程序 builder; 編碼尺 code...
  1. On the base of researching the theory of the scheme and analyzing the signal feature, it is obtained that the existence manners and character of distance information in the differential frequency signal. at the same time, a new conclusion is gained that the technology of frequency agility can decrease the constant error of system. it is also to say that frequency agility and frequency modulation fixed - distance fuze has the similar feature to random period frequency modulation fixed - distance fuze. according to the theory of address coding in the hopping - frequency communication, the paper presents the principle of selecting the frequency agility sequence which fit to the radio fuze and constructs the frequency agility sequence family based on the rs codes

    在深入研究方案原理和分析信號特徵的基礎上，獲得了該體制引信差頻信號中，距離信息的存在形式和特點，得出了頻率捷變技術的引入降低了系統定距固定誤差這一新的結論，即頻率捷變調頻定距引信在定距性能上具有類似隨機周期調頻定距的特徵。本文引入跳頻通信地址編碼理論，結合無線電引信的具體特徵，提出了適用於無線電引信的頻率捷變序列的選擇原則，並構造了基於rs碼的寬間隔頻率捷變序列族。
  2. In all kinds of complicated network, oriented linking and unlinking, communication frequency resource is strained, and bandwith to transmitting audio frequency signal is too restricted, complicated and fluky, while audio frequency data exponential have been increased in the last several years. under the circumstances, based on the research of predecessor, this paper studies wavelet analysis ' s maths gist and practices significance on signal process, and puts forward a optimized wavelet package condensation arithmetic to process audio frequency data, which gives attention to coding efficiency, multirate and compression delay. simulation experiment on the arithmetic has been done by matlab

    針對無連接和面向連接的各種復雜網路環境下，通信頻帶資源緊張，音頻傳輸帶寬有限且復雜多變，而各種音頻數據又日益增多的局面，本文研究小波分析在信號處理方面的數學依據和在數據壓縮方面的實際意義，在前人不斷工作的基礎上，提出了一種優化小波包變換編碼方案用於音頻數據的壓縮演算法，兼考慮了編碼效率、多碼率和壓縮時延多個方面，並在matlab環境下做了模擬實驗，對各種音頻信號及多種小波函數做了模擬結果比較，實驗結果證明該演算法可以在一定計算復雜度下可以很好地改進壓縮效果，達到多碼率下實現實時編解碼的過程，在高速dsp晶元等硬體設備支持下，可以有效應用於實際復雜多變信源編碼。
  3. Firstly introduced the basic theory and method with which the analog signal can be convert to digital form, including sampling theory and course, quantification and quantification error, coding, beside those we discussed some applications of sampling technology, the reason of frequency mixture and the method to eliminate it chapter 4 introduced analog mux - switch, for the reason of simpleness we only introduce it briefly

    從第3章開始，對數據採集的基本理論進行討論，首先介紹了模擬信號數字化處理中的基本理論、方法，包括采樣過程、采樣定理、量化與量化誤差、編碼，還討論了幾種采樣技術的應用、頻率混淆產生的原因及消除措施。第4章，介紹了模擬多路開關。
  4. The research on carrier spectrum, modulation schemes and signal codes of plc networks has received much attention, while the mac protocol of plc networks is usually adopted from general - media computer networks

    目前國際上注重研究plc網路的載波頻率、調制方式、信號編碼等關鍵技術。而對于plc網路的mac協議，卻通常採用修改的常規計算機網路的mac協議。
  5. In this mode 0 - 20 khz musical signals can be compression encoded. for example, it can use 32 kbits signals to compression encode 0 - 15 khz musical signals and compress stereosonic signals into 56 kbits signals. thus it can use a 64 kbits signal to provide stereosonic music of fm radio broadcasting or cds

    用該方式可將0 ~ 20khz的音樂信號進行壓縮編碼，例如，它可用32kbits的速率的信號對0 ~ 15khz的音樂信號進行壓縮編碼，可把立體聲信號壓縮成56kbits速率的信號，因而可用一個64kbits的信號提供調頻無線電廣播或相同於cd音質如身臨其境的立體聲音樂。
  6. 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。
  7. Digital speech has preponderance over analog speech in reliability, robustness and security during communication. however, digital speech needs more bandwidth than the analog signal. especially with the requirement for communication frequency increasing, it ' s necessary to code speech signal at low rates

    但是，數字化后的信號所佔的頻帶大幅增加，特別是在帶寬需求日益增長的今天，這個問題尤為突出，因此語音的低速率編碼（即壓縮編碼）成為迫切的要求。
  8. Above all, [ 12 : 8 ] harming error correction theory is mentioned in this paper. the edac circuit designed by vhdl can works normally at different frequency of the cpu clock such as 66mhz 50mhz 40mhz 33mhz. the edac function of the circuit can also be disabled by software tool. meanwhile, some basic devices such as and logic, or logic, not logic and some small scale integrated circuits are also integrated in the fpga

    本論文闡述了12 ， 8漢明碼糾錯設計過程，採用vhdl語言實現糾錯編碼器（ edac ） ，本設計能夠適應cpu時鐘信號clk2的不同頻率，如66mhz 、 50mhz 、 40mhz 、 33mhz ，並且能夠通過軟體的控制使fpga的糾錯編碼功能關閉。
  9. In this paper, the application of coherent phase - coded pulse train ( cpcpt ) solves range - doppler coupling well

    應用相參頻率編碼脈沖串信號（ cpcpt ）較好地解決了距離-多普勒耦合問題。
  10. This dissertation mainly studies the pulse compression waveform designing and signal processing, including the following aspects : firstly, by the pulse compression basic theory, the linear frequency modulation signal ( lfm ), binary phase - coded signal ( bc ) and stepped - frequency ( sf ) signal are discussed respectively including the ambiguity function, pulse compression and spectrum characteristic

    本文主要進行脈沖壓縮雷達波形設計以及信號處理方面的研究，主要體現在以下幾個方面:首先，從脈沖壓縮基本理論出發，分別給出了線性調頻、二相編碼和頻率步進信號的模糊函數、脈沖壓縮以及頻譜特性，對多普勒性能進行了簡要分析。
  11. Continuous single tune jamming is an effective jamming to linear frequency - modulated radar. however it can not suppress signal output after being full of the sidelobe suppress filter of the jammed bi - phase coded radar, no matter how great the power of it may be

    單頻正弦波干擾對線性調頻雷達是一種有效干擾，但對於二相編碼雷達無論具有多大的功率在充滿二相編碼雷達接收機的旁瓣抑制濾波器后也不能壓制信號主瓣，因此是一種無效干擾。
  12. The application of spread spectrum technology can increase dimension of signal in order to improve signal power and achieve code gain

    擴頻技術的應用可以增加信號維數以提高信號功率也可獲得編碼增益。
  13. A low bit - rate video - coding scheme based on matching pursuit is built in this section

    建立一個基於匹配跟蹤信號分解技術的低速率視頻編碼器框架。
  14. Transmition velocity relys on the style of encode and modulation essentially during modern data transmition for the quality of using line bandwidth and the immunity of code ties on them tightly. but it is important that the velocity of data transmition reaches its limitation in fact for the interface of environment and cross - talk. so for the improvement of transmition velocity, we must analyze the characteristics of noise signal and the model of line deeply and then take some useful measures to better the immunity of modulation wave

    在現代通信的數據傳輸過程中，傳輸速率本質上是由傳輸的編碼方式和調制方式決定的，因為編碼方式和調制方式直接決定了線路頻帶利用率和碼元抗干擾能力的好壞，因而直接決定了傳輸速率；但是在實際應用過程中，數據傳輸速率是不可能達到理想狀況的，因為環境干擾、串音干擾等因素的存在使得線路的頻帶不可能被完全利用起來；因此，必須認真分析線路的噪聲信號的特性以及噪聲線路的模型，以便在編碼方式和調制方式中針對性的做一些改進措施以改善調制波形的抗干擾能力，使得傳輸速率能夠進一步提高。
  15. The viterbi decoder with hard decision designed by the paper, is aimed at ( 3, 1, 9 ) convolutional coding. the data rate is 9. 6kbps. the data rate received by the rake receiver is spreaded by 127 - bit spread sequences, added pilot signals and modulated by qpsk

    該課題所設計viterbi譯碼是針對（ 3 ， 1 ， 9 ）卷積碼的硬判決譯碼，數據速率為9 . 6kbps ; rake接收機所接收的數據是擴頻因子為127 、加入導頻且經qpsk調制的擴頻信號，使用verilg硬體描述語言在xilinx公司的ise環境下在用現場可編程門陣列（ fpga ）來實現viterbi譯碼器和rake接=收機的功能。
  16. One is to use fourier transformation to convert the source signal from time domain to frequency domain and to discard high frequency harmonious components upwards of 19 ( gb / t14953 - 93 d5. 3 demanding ), then to have static huffman coding to the quantized char array which is composed of reserved direct current component and basic wave and each high frequency " s amplitudes and angles. the other is to use discrete wavelet transformation to convert the source signal from time domain to frequency domain and to set the high frequency coefficients that its absolute value is smaller than the given threshold to zero, then to have dynamic huffman coding to the quantized char array which is composed of multiple, wavelet ' s level, datum length, low frequency coefficients and reserved high frequency coefficients. mass simulinks and analyses under the two circumstances have done to show that data compression ratio is small and the relative error is also small and within the permission of engineering and the compression problem can be solved in theory of measured datum of power system

    第一種情況的壓縮方法為：採用傳統的傅立葉變換把原始信號從時間域變換到頻率域，舍棄20次及其以上的高次諧波成分（保證了gb / t14953 ? 93d5 . 3要求） ，然後對保留的直流分量、基波和各次諧波的幅值和相角數據量化后和量化時分別乘以的倍數系數構成一個數組，以字元形式保存，採用靜態huffman編碼對變換數據進行壓縮；採用離散小波變換把原始信號從時間域變換到頻率域，然後對分解得到的高頻系數進行閾值量化處理，對乘以的倍數系數、小波變換的階數、小波變換后的低頻、各級高頻以及原始數據長度、量化后的低頻系數以及保留的高頻系數大小、位置構成一個數組，以字元形式保存，採用動態huffman編碼對這個文件進行壓縮。
  17. The work of this dissertation is focused on research of some key technics in signal design and processing of mcpc signal. the main content of the dissertation is summarized as follows : it gives a detailed description of the form of the mcpc waveform and its mathematical model, and analyses the formation and properties of phase coded sequence. it discusses the effect of signal parameters on autocorrelation, power spectra and ambiguity function in forms of single pulse and pulse train and compares different kinds of single pulse signals and pulse train signals

    本文圍繞著多載波相位編碼信號在信號設計與信號處理的若干關鍵問題進行了研究，主要做了以下工作:描述了多載波相位編碼信號（ multicarrierphasecoded ，簡稱mcpc ）的信號形式，給出了其數學模型，對其中相位編碼序列的構成方式與特性進行了分析;從單脈沖和脈沖串兩種信號形式入手，對多載頻相位編碼信號的參數在自相關函數、功率譜密度和模糊方程上的影響作了詳細地討論，對不同調制方式的單脈沖和脈沖串信號進行了比較。
  18. Telecommunications - digital processing of program audio signals - algorithm for 15 - khz audio at 384 kbit s using 14 11 bit coding

    電信.聲頻信號程序的數字處理.使用14 11位編碼以384k位秒速率傳輸15k赫茲聲頻的演算法
  19. Based on the statistics characteristic of prc - phase - modulation cw phase - array guidance radar, this thesis discusses the signal processing method and cfar algorithm of multiple repetition frequency pseudo - noise m - sequence phase coded signal

    本文結合偽碼調相連續波信號體制的特點，深入研究了多參差重復頻率偽隨機m序列相位編碼信號的處理方法和恆虛警檢測演算法。
  20. System scheme of speech coding plus spread spectrum communication was presented based on a full analysis of noise characteristic, attenuation characteristic and impedance characteristic of low - voltage power line. spread spectrum carrier ( abbreviated as ssc ) technology is adopted to overcome problems existing in signal transmission over power line. high quality, low rate mbe compression algorithm was used to complete speech encoding and decoding

    在對低壓電力線路的噪聲特性、衰減特性和阻抗特性三個方面充分分析的基礎上，本文提出一種語音編碼+擴頻傳輸的系統總體方案，採用擴頻載波（ spreadspectrumcarrier ，縮寫為ssc ）技術克服電力線傳輸信號存在的問題，採用語音合成質量高並具有較低碼率的mbe壓縮演算法完成語音信號的編解碼。
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