比特號碼 的英文怎麼說

中文拼音 [bǐtèháomǎ]
比特號碼 英文
bit number （bn）
  • 比 : Ⅰ動詞1 （比較; 較量高下、 長短、距離、好壞等） compare; compete; contrast; match; emulate 2 （比...
  • 特 : Ⅰ形容詞（特殊; 超出一般） particular; special; exceptional; unusual Ⅱ副詞1 （特別） especially; v...
  • 號 : 號Ⅰ名1 （名稱） name 2 （別號; 字） assumed name; alternative name3 （商店） business house 4 （...
  • 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...
  • 比特 : [計算機] bit （信息量單位）比特波形 bit pattern; 比特差錯率 bit error rate [probability]; 比特緩...
  1. Tootle does not have a listed phone number, and jail and sheriff ' s records do not list an attorney for her

    電話本上沒有屠特的電話號碼，監獄和艾斯坎比亞縣也找不到她的律師紀錄。
  2. The performance targets of acquisition and tracking : the code rate of pseudorandom signals is 1m - 5mhz, the code length is 255. 511, the receiving sensitivity is - 118dbm, the dynamic range is 70db. the doppler of carrier is 75khz, the time that 10 targets are captured within 0. 2s

    捕獲和跟蹤的性能指標：偽隨機信號碼率1m 5m比特/秒，碼長是255 、 511 ，接收靈敏度是- 118db ，信號動態范圍是70db載波普勒是75khz ，目標捕獲在0 . 2秒內。
  3. The adaptation processing includes linear prediction coefficient adaptation and adaptation of quantization step size for residual signals. based on g. 726, we adopt a huffman coder to make use of probability statistic of bit cascade covering n ( n 1 ) samples generated from adpcm, in order to further reduce the bit rate. ng is lossless entropy coding, the speech quality of our improved algorithm should be same as that of g. 726 standard

    我們的研究和改進工作包括:研究最優非均勻自適應量化器，及其自適應演算法;研究波形預測函數，以及函數零點、極點的自適應演算法;基於每n （ n 1 ）個樣本所對應符號的概率統計，對預測殘差量化值再進行huffman編碼，進一步降低比特率。
  4. In this paper, chroma dc coefficients are selected as the carrier data because chroma dc coefficients are robust. also, alterable steps are used to select one part of the coefficients, so the watermark is imperceptible ; before watermark embedding, the watermark is divided into many parts, every part is individually embedded into one gop of the video, even if a gop is destroyed, the watermark can be extracted correctly, this methods promotes the robustness of the watermark ; in addition, in order to promote the security of the watermark, the user ' s id and password are used to generate chaos sequence by the chaos system which is created in this paper, later, watermark is mixed by the chaos sequence. also, the embedding position of the watermark bit is modified by one chaos sequence, so, unauthorized person can not extract or remove the watermark, since the embedding position is unknown

    本文認為，色度dc系數是魯棒性非常好的參數，因而選擇色度dc系數作為水印信息載體，同時，採用可變的步長選擇部分系數，保證了水印的隱形性；在嵌入水印時，本文採用水印信息「網格劃分」 、各子塊獨立嵌入視頻的方案，由於水印信息子塊是相對獨立的嵌入視頻中的每一相對獨立的圖組當中，即使某一圖組收到一定破壞，也能夠恢復水印信息，使水印的健壯性得到提高；此外，為了提高水印信息的安全性，在嵌入水印信息時，根據用戶輸入的id號和密碼，利用本文構造的混沌系統產生的混沌序列對水印信息進行變換，同時，對每一水印信息比特的嵌入位置也採用了偽隨機序列進行調整，這樣，未授權用戶不能提取水印信息，也難以擦除其中的水印信息，因為嵌入的位置是未知的。
  5. This paper mainly discusses the design principles and chief techniques of a digital accessing system for power - line communication net ( plcn ). the technology of low bit rate speech compression high - speed modem based on plcn adaptive equalization to the channel anti - jamming and anti - fading are applied in this system. so speech tele - control data and tele - protection signals can be transmitted high quality in the band - limited channel

    該系統綜合應用了低比特率語音信號壓縮編碼技術、基於電力通信網的高速調制解調技術、信號傳輸的通道自適應均衡技術和抗干擾、抗衰減技術，可在帶限通道中高質量的傳輸語音、遠動數據和遠方保護等信號，具有較高的整體性能。
  6. Most of these standards are based on the method of inter - frame motion compensation and two - dimensional discrete cosine transform ( 2d - dct ) and encode and describe the color video in ycbcr 4 : 2 : 0 format, which want to take advantage of human visual system ( hvs ) to save bit expense by decreasing the resolution of two color difference components

    當前國際上的壓縮標準普遍採用幀間運動補償加幀內二維離散餘弦變換的編碼方法，並且將彩色視頻序列表示為ycbcr格式，試圖利用人眼的視覺特性降低對色差信號的解析度來節省比特開銷。
  7. By reducing coding rate, more speech signals can be transferred in the same channel. so, low bit rate speech coding has especially important significance when the transmission rate is limited very strictly

    通過降低編碼速率，可以使同樣的通道容量能夠傳輸更多路的語音信號，在傳輸比特限制十分嚴格的場合，低速率語音編碼具有特別重要的意義。
  8. Operator, i want to make a person - to - person call to miss. marilyn peters in washington d. c. the number is 393 - 5121

    接線員，我想給住在哥倫比亞特區華盛頓的瑪麗琳.彼得斯打個直呼電話，號碼是393 - 5121 。
  9. " if you ' re making a 30 - second call every morning at about 9 a. m. and the number does n ' t match those used by your colleagues then we can guess fairly accurately that ' s your spouse, " said robert picton, product manager at south african it firm dimension data

    南非it公司dimension data的產品經理羅伯特比克頓說： 「如果你每天早上9點鐘左右打一個30秒的電話，而且你所撥打的號碼不是同事們的常用號碼，那麼基本可以確定，這個電話是打給情人的。 」
  10. G723 aerithemetic is a compressing arithemetic that proposed by itu - t and applied in speech and other audio frequency signals of low velocity multimedia services, such as : h. 323, h. 324 system. this arithemetic provides inspection to silence speech frames and fills in comfortable noise when it is silence. if optimize system and increase the complexity limitedly, we can get higher quality of speech. g723. 1 is also available in music or other voice signals, but the managing effect is not as good as speech ' s

    G . 723演算法是itu - t建議的應用於低速率多媒體服務中語音或其它音頻信號的壓縮演算法，例如： h . 323 ， h . 324系統。這種聲碼器具備兩種比特率： 5 . 3kbps ， 6 . 3kps 。在幀邊界處可以在兩種速率之間進行切換。
  11. Delivering quality video over wireless channels in real time is a challenging task. this isprimarily because of the throughput of a wireless channel may be reduced due to multipath fading, cochannel interference, and noise disturbance. therefore, there is a critical need forrobust transmission of video over wireless channels. recently, the emergence of mimo - ofdm has stimulated great interest in real time video communication, because this system can offer broadband for the multimedia data transmission over wireless channels

    因為傳統的視頻壓縮技術是針對單天線系統，只產生單一的比特流，這與多天線系統不相適應，為此，需要將視頻信號分解成相應的多個碼流。本文探討一種將能視頻信號分解成多個碼流的多描述編碼方法，以適應多天線系統的傳輸。
  12. The paper makes great efforts on the software optimization of evrc vocoder. based on the understanding of tms320c64xx cpu structure, we do deeply - optimization on the loop which appear usually in voice signal processing, and this improve the utility ratio of cpu and the parallelity degree of cpu function cell. at the same time, we utilize the bit - exact test to test the fixed - point evrc vocoder with the test vectors of tia / eia / is - 718, which improve the robustness of the vocoder

    本文圍繞定點evrc聲碼器的軟體優化，做了大量的工作，在充分理解tms320c64xxcpu結構的基礎上，針對語音信號處理中大量出現的循環運算進行了深度優化，大大提高了cpu的利用率以及cpu功能單元的并行程度，同時，我們還用tia / eia / is - 718的測試向量對定點evrc聲碼器進行了嚴格比特對準測試，提高了聲碼器的魯棒性。
  13. " he said if he was going to go down, he was going to go down in larry bird ' s jersey, " oklahoma county district judge ray elliott said wednesday. " we accommodated his request and he was just as happy as he could be

    俄克拉荷馬縣地區法官雷埃利奧特19日表示： 「托爾比指出，如果要去坐牢，那麼他希望能按照拉里伯德的球衣號碼來決定自己的期限。
  14. Based on the criterion of maximizing the total average signal to interference plus noise ratio ( sinr ), the optimal subchannel allocation strategy is obtained in theory when random signature sequences are used. for its implementation, an iterative algorithm is proposed, which is similar to the water filling principle. by using the proposed algorithm, we obtain significant improvement on the performance of the system, and the transmission quality can be guaranteed

    根據最大信號干擾加噪聲比原理，分析得出了隨機特徵碼條件下最優的子通道分配策略，並提出了一種類似於灌水原理的子通道分配演算法，該演算法能在保證各個用戶傳輸質量的前提下，使系統的整體性能接近最優。
  15. The g. 726 speech coding algorithm involves adaptive linear prediction to eliminate the correlation of signal waveform, so as to reduce the resulting bit rate

    G . 726建議採用自適應預測編碼，以消除波形信號的相關性，從而降低語音信號表達的比特率。
  16. A modified synchronization approach by using sliding windows is proposed. the pre - processing for obtaining principal eigenvectors is a smoothing procedure for the dsss signal. the computer simulations show that the approach is more convenience in low snr comparing with other code synchronization methods

    3 、提出了一種改進滑動窗的盲同步方法，該方法利用求主特徵向量的過程，對擴頻信號進行平滑降噪預處理，實現了較低信噪比下的碼同步。
  17. A novel and generic approach is presented to the hardware implementation of the rsa cryptoprocessor in deep submicro technology with a redesigned systolic array

    長比特1024位以上數據的全局信號傳輸和乘法器的動態分割問題，對于rsa密碼處理器的速度提高是非常重要的因素。
  18. In enhanced fm - dcsk technique, instead of transmitting only one information - bearing signal after the reference signal, n bits will be transmitted using the same reference and the date rate is increased

    改進的fm - dcsk系統是在每一個參考信號的碼片后，連續傳輸兩個或多個信息比特，增加了傳輸速率，使得fm - dcsk系統的性能得到了進一步的改善。
  19. This chip integrates the algorithms of coding and decoding in chip, which can accurately control the compression rate and compress many video signal formats, including pal and ntsc. on the other hand, it can also obtain excellent image quality and high compression rate. in this paper, we make the best use of adv6oilc fulfill the remote multimedia surveillance, which provides many advantages in implementing such system

    該專用壓縮解壓縮晶元系列片內集成了編解碼演算法，具有精確的壓縮比特率控制，能實時對包括pal和ntsc在內的多種視頻信號進行壓縮和解壓縮，同時由於採用小波變換進行圖像編解碼，能保證獲得很高的壓縮率和圖像質量。
  20. Meanwhile, the telephone gateway in tetra system is introduced. in further research, the principle of tetra speech coding algorithm ? algebraic codebook excitation linear prediction ( acelp ) is introduced and analysed in detail, which is a advanced codebook excitation linear prediction ( celp ). acelp algorithm replaces the excitation signals with algebraic codebook and uses some technique such as minimizing the mean square error ( mse ) and the analysis - synthesis method to obtain characteristic parameters of speech

    同時，介紹tetra系統的市話網關，並在接下來的研究中詳細介紹tetra電話網關中應用到的語音編解碼演算法? ?代數碼本激勵線性預測碼（ acelp ）的基本原理，它是一種簡化了的碼本激勵線性預測碼（ celp ） ，它把激勵信號用代數碼本代替，並且運用了均方誤差最小、分析?合成等技術提取出語音的特徵參數，極大地降低了比特率，而且具有較好的重建語音質量。
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