時間頻率編碼 的英文怎麼說

中文拼音 [shíjiānbīnlǜbiānmǎ]
時間頻率編碼 英文
time-frequency code
  • 時 : shí］Ⅰ名1 （比較長的一段時間）time; times; days:當時at that time; in those days; 古時 ancient tim...
  • 間 : 間Ⅰ名詞1 （中間） between; among 2 （一定的空間或時間里） with a definite time or space 3 （一間...
  • 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...
  • 時間 : time; hour; 北京時間十九點整19 hours beijing time; 上課時間school hours; 時間與空間 time and spac...
  • 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. Therefore, during system design, cascade coding combining reed - solomon codes and convolutional codes is utilized ; in addition, time - direction and frequency - direction interleaves are added to alleviate channel fading

    因此在進行系統設計時，採用了rs碼與卷積碼的級聯編碼方案，並同時加入時間交織和頻率交織來對抗通道衰落。
  2. The motion - based layered compression scheme in this dissertation is different from the traditional full picture - process method, using different encoding strategy by classify marcoblock in temporal motion, a high encoding efficiency is achieved in experiment

    本文提出的基於運動的分級壓縮演算法，打破了以往時空伸縮全幀處理的概念，實驗中以塊為單位按時間頻率分等級，分別採用不同的編碼策略，達到較好的編碼效率。
  3. 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。
  4. The article originated some conversion algorithms based on the data flow of multiplexing. it includes de - multiplexing ps, pretreating audio, disposing ps / ts system layer, calculate time information, controling stream output. the project describes the five parts of software system

    本文分析了轉碼設計中的時間信息、碼率等關鍵問題，提出了按碼流合成的數據流程進行轉碼的架構，實現了編碼演算法: ps流解復用、音頻預處理、 ps / ts的系統層處理、時間信息處理、碼流輸出控制共五個部分。
  5. This thesis analyzed the error performance of mfsk and mpsk modulator in hf with optimal receiver, studied the shortcoming and strongpoint of several typical short - wave data transmission systems and their performance over fading and intersymbol interference channels. the theory and key technical of adaptive modem is discussed. in the last, an efficient modulation scheme - block coded modulation is introduced into hf radio systems to improve the efficiency of hf data transmission, this paper also proposes a hf data transmission system scheme composed of a block coded modulator and a decision feedback type adaptive equalizer

    本文在對短波數傳兩種常見調制方式fsk及psk最佳解調性能分析的基礎上，討論了幾種採用典型調制技術的短波數傳系統（如時頻調制、多進制頻移鍵控慢跳頻、 chess系統）原理、優缺點以及抗衰落和抗符號間干擾的性能；分析了自適應串列數傳系統抗短波通道衰落和多徑干擾的原理、關鍵技術；在本文的最後引入一種有望解決短波數傳系統低效率狀況的調制方式bcm -分組編碼調制，給出了一個採用bcm技術與自適應均衡技術相結合的短波數傳系統方案，並通過計算機模擬進行了初部驗證。
  6. In this thesis, first, we present the theory of sess system, the generation of the sess spreading code and its characteristics and the acquisition theory of conventional spread spectrum communication system. an efficient acquisition scheme based on periodically transmitting the synchronization head, which is composed of binary chaotic codes, using the matched filter and automatic decision threshold - level control based on a so - called constant false alarm criterion for sess system is present. the acquisition model of sess system is built and simulated in the awgn channel, the raleigh fading channel and imulti - address interfere condition

    本文首先概述了自編碼擴頻通信的原理、自編碼擴頻序列的產生方法及其特性和擴頻通信系統編碼同步的理論，然後針對自編碼擴頻通信系統提出了擴頻序列捕獲方案:周期性地加入混沌序列同步碼，並採用恆虛警率匹配濾波器捕獲法;在加性白高斯噪聲通道、瑞利衰落通道和多址干擾情況下進行了模擬，分析了各種捕獲性能:在選擇性能最優的混沌序列、適當的序列長度、虛警概率及門限值的情況下，可以獲得較短的捕獲時間和較大的捕獲概率。
  7. The experimental results manifest that the psnr of both the full reconstructed video sequences and the ones of frame rates changed are improved compared to the system base on haar wavelet. moreover, the decoding time of db2 filters is shorter than the haar one, the shorter the decoding time, the better the real - time displaying in the client side

    實驗結果表明，本系統與基於haar小波的可伸縮性視頻編解碼系統相比，完全重建的視頻圖像質量和幀速率改變后的重建視頻圖像質量都有了明顯的提高，並且解碼時間也有所縮短，具備了更好的實時性。
  8. The characteristic and key technologies of the system are as follows : ( 1 ) in realizing the live broadcast of audio and video, the problem of immense multimedia data and low networks bandwidth utilization ratio is solved by using mpeg - 4 as format of audio and video data. audio and video data are collected by video card cv500 which developed by beijing sum tone company ; meanwhile, the contradictory between the delay of networks transmitting and the quality of the image is well solved by setting a " bi - buffer area "

    系統實現中解決的關鍵問題和特色主要有以下幾個方面： （ 1 ）在視音頻直播功能的實現中，通過使用北京算通公司的cv500視頻採集卡和cv500sdk進行視音頻數據採集，並採用當今最新的圖像和語音編碼壓縮標準mpeg - 4作為視音頻數據的採集格式，既保證了圖像的質量，又大大縮減了視音頻所佔的帶寬，從而解決了多媒體數據量大、網路帶寬利用率低的問題；同時，通過設置環形緩沖區的辦法來調和網路傳輸延時與圖像質量之間的矛盾，取得了較好的效果。
  9. 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。
  10. Based on the analysis of image wavelet transformation and the space / frequency distributing characteristics of different subbands " coefficients, this dissertation fully exploits the following theories and methods : scalar quantization, vector quantization, trellis coded quantization, trellis coded vector quantization, vector classification, codebook expansion and weighted mean square error rule basing mankind visual characteristics, etc. from different angles of information amalgamation, it develops several innovative algorithms of image compression and coding, gives their realization schemes, and makes plentiful simulation tests

    本文在分析了圖像小波變換的原理和子帶系數空間及頻率分佈特點的基礎上，充分利用標量量化、矢量量化、網格編碼量化、網格編碼矢量量化、矢量分類、碼書擴展和基於人眼視覺特性的加權均方誤差準則等思想和方法，從信息融合的不同角度展開了對小波圖像的壓縮編碼研究，同時也討論了這些方法在靜止圖像量化中的具體應用。
  11. Space - time coding technology is based on multi - elements antenna array and has strong points of both time processing and space processing. it can satisfy the requirements of high data rate transmission in the future radio mobile system without the increase of the transmitting power and the spread of the frequency. and it can greatly improve the quality of communication services and system capacity under fading channels

    空時編碼技術是一種利用多天線陣發送和接收的技術，該技術能很好地結合時間處理技術和空間處理技術的特點，能在不增大發射功率和不擴展頻帶的前提下滿足未來無線移動通信對高數據率傳輸的要求，從而全面提高衰落通道的通信質量和通信系統容量。
  12. Because of the affection of noise and signal fading, in order to high data rate access and high quality, wireless communication need new technology to improve the link reliability and enhance the spectrum efficiency. mimo ( multiple input and multiple output ) can enhance the spectrum efficiency and increase channel capacity greatly, and reduce the multipath affection without spectrum band and power increasing. the 3th generation mobile communication wcdma is coming, with the purpose of putting mimo, stbc ( space time block coding ), space diversity into practice of wcdma systems, in order to lay the foundation of 3g toward 4g, this dissertation has done some research under this background

    在發射機和接收機使用多個天線進行數據傳輸的多輸入多輸出（ mimo ）技術，可以在不增加帶寬和天線發送功率的條件下，成倍的提高頻譜利用率，提高系統的通道容量，還可以抗多徑干擾。第三代移動通信wcdma即將到來，為了將多天線技術（ mimo ） ，空間分集技術，空時編碼技術用到wcdma系統中去，為第三代移動通信向第四代移動通信系統平穩過渡奠定一定的基礎，需要做一些理論上的研究。
  13. Generally speaking, a coding algorithm must be low complexity and work at high speed. in this paper, the coding methods are studied to increase coding speed. firstly, a novel method based on temporal prediction to detect all - zero dct coefficients is proposed, which can greatly reduce the coding cost and maintain a good tradeoff between correct detecting ratio and false detecting ratio with little additional computation

    本文以h . 263壓縮編碼標準為基本框架，針對其中影響壓縮效率和速度的瓶頸展開研究，包括快速運動搜索演算法的研究、預判零方法的研究以及mmx技術在視頻編碼中的應用等，提出了一種基於時域預測的預判零新方法，在不影響重建視頻質量的條件下具有較高的判別效率，節約了大量dct變換和量化的時間，提高了編碼的速度，達到了實時性要求。
  14. Because using ofdm, a frequency - selective fading channel can be transformed into multiple frequency - flat fading sub - channels, in the coding process, spatial diversity ( due to multiple antennas ) is integrated with frequency diversity ( due to delay spread ), therefore, the proposed code can provide higher diversity gain than space - time code alone

    該編碼利用正交頻分復用把頻率選擇性衰落通道變換成平衰落通道的特性，同時引入了基於多天線的空間分集和基於時延擴展的頻率分集，比空時編碼具有更高的分集增益。
  15. Stfs is a good coding arithmetic which is simple and easy to realize. moreimportantly stfs has higher diversity order and encoding gain than sts at the samesymbol rate. it can repel channel fading, improve data rate and wireless systemcapacity availability, stand against frequency selectivity fading, eliminate theintersymbol interference

    該編碼演算法不僅在方法和原理上簡單，而且易於實現，在相同符號率下比空時擴頻具有更大的分集度和更高的編碼增益，能有效抵抗通道衰落、提高數據速率、提高無線系統容量、很好的對抗頻率選擇性衰落和消除符間干擾。
  16. In term of bandwidth fluctuation of internet, proposes two types of scalable video coding scheme adapted to transmission of internet, that is, a fine - granularity spatially scalable video coding and a progressive fine - granularity scalable video coding. meanwhile, proposes a wavelet transform ' s algorithm using for lower rate coding

    同時針對internet上傳輸帶寬波動性，提出了兩種可用於internet傳輸的可伸縮視頻編碼方案- - fgss （精細空間可伸縮視頻編碼）和pfgs （漸進精細可伸縮視頻編碼） ，提出了用於極低碼率的小波變換編碼演算法。
  17. Space - time block codes ( stbc ) based on orthogonal design has attracts enough attentions in its applications in mimo - ofdm systems for its full diversity exploiting and low coding / decoding complexity. these applications include not only system performance improvement, but also reduction of the peak to average power ratio in mimo - ofdm systems. under these backgrounds, this paper first studies space - time block codes under the single carrier, flat fading situation, discusses the encoding / decoding principle and its performance under different channels. then, the author focused on the performance of space - time codes under time selective and frequency selective channels, uncovered the fundamental cause of how maximum doppler shift and power delay profile influences the performance of space - time codes through deduction and simulation

    在上述背景下，本文以單載波，平坦衰落通道下的空時分組碼為切入點，深入分析了其編譯碼原理和在不同通道情況下的性能;以此為基礎，後文展開了對空時分組碼在多載波，時間選擇性和頻率選擇性衰落通道下的性能研究，通過理論推導和充分的模擬，揭示了空時/頻分組碼在時頻選擇性衰落通道中的性能變化的根本原因;在以上對多載波系統中空時分組碼的研究的過程中，作者發現了空時分組碼在降低系統峰平比方面應用的可行性，提出了一種降低峰平比的新方法。
  18. In this paper, we propose several techniques to improve heterogeneous transcoding performance, which is transcoding from mpeg - 2 to mpeg sp @ ml. bit - rate, frame rate and spatial resolution transcoding are all studied here. first, we compared the difference between mpeg - 2 and mpeg - 4

    本文以mpeg標準間視頻流轉碼為背景，主要研究了mpeg - 2到mpeg - 4sp @ ml （簡單框架主級， simpleprofile @ mainlevel ）的轉換編碼，其中包含了碼率、空間解析度和時間解析度轉碼的內容。
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