編碼率 的英文怎麼說

中文拼音 [biānmǎlǜ]
編碼率 英文
code rate
  • 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...
  • 率 : 率名詞（比值） rate; ratio; proportion
  • 編碼 : encoded; code; coded; encrypt; codogram; coding編碼表 encode table; 編碼程序 builder; 編碼尺 code...
  1. Different bit rates are allocated to wavelet blocks according to energy the wavelet blocks include. bit rates in wavelet blocks are adaptively adjusted in coding process. by use of bipartition, the entropy of each wavelet block approximates to the target bit rate of one

    該演算法首先根據每個小波塊所含能量的多少和到每個小波塊實際編碼所用的比特數，給其分配不同的碼率；然後根據二分法，通過調整各小波塊的量化因子使得各小波塊的熵逼近它的目標比特率。
  2. The hardware of the ip phone codec to be designed is based on the fixed point digital signal processor ( ti ' s tms320vc5410 ) while the compression and decompression core in the software of dsp is based on the itu - t vg. 729a. ip phone codec carryout the task of collecting / playing - back. coding / decoding of speech signal and communication with embedded cpu. etc

    該語音編解碼器的硬體基於tms320vc5410 ，編解碼演算法遵循itu - tg . 729a協議，能夠實現語音信號的採集/回放、編碼/解碼以及同嵌入式cpu通信等功能，在8kbit / s的碼率下能夠提供獲得良好的語音質量。
  3. 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。
  4. Lc apparatus almost meet all the needs of space optical communication such as weight, size, power consume, life, cost, driving voltage, intergration of optics and electricity, programe, optically take ? over aperture, beam scanning, deflexional range and so on. switches, deflexional facilities and scanning equipments which made with lc have been used in the system of labor in space communication. the only bug of lc apparatus is that their answer speed only get microsecond rate or submicrosecond rate. but it is practical for them to be used in special beam capture, scan, deflexion controling which don ’ t concerned with code rate and code type

    液晶器件幾乎滿足空間光通信的所有大的指標要求如重量、尺寸、功耗、壽命、成本、驅動電壓、光電集成、可編程性、光學接收孔徑、光束掃描和偏轉范圍等等。液晶光開關、光偏轉器、光掃描器已經開始應用於光纖通信的實驗系統中。液晶類器件應用於光通信的唯一重大缺陷，是其響應速度目前只能達到微秒級或亞微秒級，不過，在不涉及到碼型碼率的空間光束捕獲、掃描、偏轉、控制方面，液晶器件完全可能進入實用化。
  5. This scheme firstly determined the operating mode of coded video stream in nal layer according to the current channel state before they were packed. the coded video stream operated under ssm in error - free channel so as to reduce the packing tradeoff and increase the coding efficiency. while in error - prone channel, they operated under dpm combined with improved unequal error protection ( uep ) scheme based on human eyes characteristic in order to elevate their robustness to channel error

    該方案中編碼視頻信息在進行打包封裝前，首先根據當前通道狀態信息自適應地確定其在nal層中的工作模式，在無丟包通道中採用單數據片模式以降低編碼視頻流的打包開銷，提高編碼效率；而在丟包通道中採用數據分區模式以提高編碼視頻流對通道誤碼的魯棒性能，同時基於人眼視覺特性，提出了一種改進的uep策略。
  6. Ajl those drictions supp1ied by thes system help us resolve the problern of managing aluxninuin type materials drawings. at the sazne hme, it makes the exploitation and design ofproduc has better inherit by utilising the functions of coding based on the software. as thes systern app1ied the classiforg and coding prinop1e based on gt and cad tedm1ogy in design and managernen of drawings which not only enhance the design quality of product, shorten the development pedod but also decrease the ropeaed drawing work and hanme the method of protract drwings

    該系統規范了型材圖紙的設計，提高了型材圖紙的檢索效率，對型材的相關信息進行了的效的管理，良好地解決了鋁型材圖紙檢索和管理的問題；同時，作為一個計算機輔助產品設計系統，利用該系統的分類檢索功能和圖紙支撐軟體的圖形編碼功能，使產品的開發設計具有良好的繼承性。
  7. For the anticounterfeiting of printings ( such as certificate ), the existent many ways ( such as rainbow holograms ) are not available as the need for special use : anti - distortion and anti - copy. basing in the double - random - phase transform, this article puts forward a new way that two - dimensional bar code is used as anticounterfeiting label with anti - printing ? scanning and anti - damage properties. the major job are : ( 1 ). basing in the ascii codes, numerals and alphabets are encoded and subsequently transformed into two - value bar code matrix figure. later, using amplitude - based double - random - phase transform, the enciphered gray scale figure is formed. by computer simulation ( 4f system ) and printing as well as scanning ( 20 times ), we get the result that the gray scale figure with little miscoding rate ( 0. 0026 ) by “ matrix expanding way ”

    本文主要開展了以下三項研究工作: （ 1 ） .用自定義編碼方案，將數字和字母（ ascii碼）轉換成二值條碼圖;對該圖形進行振幅型雙隨機相位加密變換，得到原信息的加密灰度圖;通過4f光學系統計算機模擬和列印-掃描實驗，證明本文提出的「矩陣擴展法」灰度圖具有較小的誤碼率，對於20次列印-掃描實驗，誤碼率不大於0 . 0026 。
  8. In this paper, combined with currently voice coding technique, espically with the fabulosity development of the mixed voice coding and the increasingly utility of the digital signals processor. we investigated the voice coding technique and discussed emphasizedly the technology of variable rate voice coding technology

    本文結合當前語音編碼技術尤其是混合編碼技術的驚人發展及數字信號處理器的日益實用化，研究了語音編碼技術，並重點討論了變速率語音編碼技術。論文簡要介紹語音編碼技術中的波形編碼和聲碼器的主要性能。
  9. 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。
  10. 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。
  11. 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。
  12. The focus of this paper is the problem of frequency optimum codes of irrelevance processing. the method of researching optimum codes by using costas array method is given

    著重討論了非相干處理時的頻率最優編碼問題，給出了以costas陣列方式尋找最優編碼的方法。
  13. As a combination of ofdm with space - time coding technique, mimo - ofdm becomes more and more important in the future wireless communication systems, the mimo - ofdm system can not only effectively enhance the transmission rate and capacity of the wireless communication system but also greatly mitigate the effections of mufti - path fading and interfere

    Mimo - ofdm技術將ofdm與空時編碼技術有機的結合在一起，能夠大幅度的提高無線通信系統的通道容量和傳輸效率，並能有效的抵抗多徑衰落、抑制干擾和噪聲。
  14. 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。
  15. Code this control code of the recording s bit inside of the 16 the instruction to tell to put in the hdcd in the cd hdcd in the phonograph solution code, is central plains the number this of the high of that short restores out. like this, hdcd can in the cd of 44. 1 orotund number that khzs sampling frequency extreme limit inside, exceed the bandwidth 20 khzs re - appeared out

    在hdcd編碼錄音的第16bit中的這個控制代碼指令告訴放在cd唱機中的hdcd解碼器，把訊號中原本的那個短促的陡高音還原出來。這樣， hdcd就能夠在cd的44 . 1khz的取樣頻率極限內，把頻寬超過20khz的聲音訊號重現出來了。
  16. 2, under equally see the quantity, compress the calculate way s exportation code plain deal connect the good and bad that decide its function of the good and bad, or say, output the code the term of same alike rate the bottom, and the portrait quantity reflected the function of the compression calculate way. adopt with dv all because of the m - jpeg an inside compress the way, efficiency of their compression certainly want lower than mpeg2. certainly, this is an average circumstance, at the time that low code rate, the mpeg2 can compare the m - jpeg compression the ratio high and a lot of but keep the good diagram to resemble the quantity ; but at request the diagram to resemble the quantity the good time for example the sow the room program edit with empress period creation, their difference to is not very big. too is to say, just at the compression efficiency that not emphasize the portrait quantity that the circumstance, mpeg2 is high

    2在同樣的視頻質量下，壓縮演算法的輸出碼率直接決定其性能的優劣，或者說，輸出碼率相同的條件下，圖象質量的優劣反映了壓縮演算法的性能。由於m - jpeg和dv都是採用幀內壓縮方式，它們的壓縮效率當然要比mpeg2低。當然，這是一個平均情況，在低碼率的時候， mpeg2可以比m - jpeg的壓縮比高很多而保持較好的圖像質量而在要求圖像質量很好的時候比如演播室節目編輯和後期製作，它們的差別不是很大。
  17. Needless to say fixed quantizer / quality has very limited use for most of us

    所以說定比特率編碼對我們的用處很小。
  18. In the proposed method, the controller takes the buffer length as congestion indication, takes sources quality and bandwidth utility as object function so as to learn on line. as the controller outputs, the coding rate for input traffic sources and the corresponding user percentage are used to adjust the cells " arrival rate to the multiplexer buffer. compared with the previous method where cells " arrival rate is tuned only by the encoding rate and the encoding rates for all input traffic sources are regulated in a body, the proposed method guarantee that the quality of cells are optimal while cell loss rate is minimized, which means quality of service is guaranteed

    在該方法中，擁塞控制器以緩沖區大小信元作為擁塞指示，以信源質量和帶寬利用率作為目標函數進行在線學習，控制器輸出包括信源編碼率及其對應的用戶數在全部用戶中所佔的百分比，即根據信源編碼率及對應的用戶百分數調整信源輸入流，從而克服了以往擁塞控制方法中僅僅調整編碼率帶來的對所有信源進行整體調整的缺陷，使控制系統在信元損失率最小情況下確保信源輸入流質量最高，從而有效地利用了網路帶寬。
  19. Voice encoders may be further enhanced by encoding digital signals at variable encoding rates

    語音編碼器可以通過在可變編碼率編碼數字信號進一步加強。
  20. The project uses for reference the algorithm thought of sbc ( subband coding ) to measure off the audio to the corresponding frequency width and encode it by the different sensitivity of human hearing, which results in the lower coding rate and bearable voice quality. the algorithm processing low bit - rate audio is designed to be self - adaptive by the situation of network. the component developped by that algorithm and project has already been used in the realtime interactive educational system

    該方案借鑒sbc （ subbandcoding ）子帶編碼演算法思想，將音頻按對人聽覺敏感程度不同劃分為相應的頻帶並進行相應的編碼，從而得到較低的編碼率和較好的語音質量，設計了可根據網路狀況進行自適應的低帶寬音頻處理演算法。
分享友人
分享到 Line

分享到臉書