coding wave 中文意思是什麼

coding wave 解釋
編碼波
  • coding : n. 編碼；譯成電碼。
  • wave : n 〈美海軍〉女志願軍人〈見 WAVES 條〉。n 1 波浪；碎浪； 〈the wave（s）〉 〈詩〉海。2 波動；波狀...
  1. It synthesizes the excellence of wave coding and parameter coding, adopts vector quantity, analyse - synthesize, perceptual weighting, therefore, gains good speech coding quality at 8kbit / s. cs - acelp can be used in individual telecom, iphone, c / n, microwave telecom and isdn

    Cs - acelp演算法綜合了波形編碼和參數編碼的優點，以自適應預測編碼技術為基礎，採用了矢量量化、合成分析和感覺加權等技術，在8kbit / s速率上獲得了較高的語音編碼質量。
  2. 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。
  3. It brought forward genetic algorithms with binary character string coding, genetic operation is the best optional optimization preserving strategy operation, multipoint crossover and nonsymmetrical mutation based on researching widely the genetic algorithms " characteristics of varied coding modes, selection modes, crossover modes and mutation modes in allusion to genetic algorithms is difficult of multi - parameter coding and genetic operation ' s realization for wave impedances inversion

    針對遺傳演算法用於波阻抗反演涉及多參數編碼、以及對應遺傳操作不易實現的問題，在廣泛研究了遺傳演算法的各種編碼方式、及其對應的選擇方式、交叉方式以及變異方式特點的基礎上，提出了採用二進制字元串編碼、遺傳操作為最優保存策略選擇、多點交叉和非均勻變異的遺傳演算法。
  4. This paper brought forward global optimized wave impedances mixed inversion based on genetic algorithms with binary character string coding, genetic operation is the best optional optimization preserving strategy operation, multipoint crossover and nonsymmetrical mutation based on researching widely the genetic algorithms ' characteristics of varied coding modes, selection modes, crossover modes and mutation modes in allusion to genetic algorithms is difficult of multi - parameter coding and genetic operation ' s realization for wave impedances inversion

    摘要針對遺傳演算法用於波阻抗反演涉及多參數編碼、以及對應遺傳操作不易實現的問題，在廣泛研究了遺傳演算法的各種編碼方式、及其對應的選擇方式、交叉方式以及變異方式特點的基礎上，提出了採用二進制字元串編碼、遺傳操作為最優保存策略選擇、多點交叉和非均勻變異的遺傳演算法，基於該演算法形成了全局尋優的波阻抗混合反演方法。
  5. Drm is a technology aims to improve sound quality of am transmission below 30 mhz and to reduce interference common in medium wave environment. new compression technologies known as mpeg4 advanced audio coding ( aac ) and spectral band replication ( sbr ) are employed to offer sound quality comparable to that of fm transmission

    它是針對30mhz以下的調幅（ am ）廣播而設計，大大提高了音質，並改善了中波廣播較易受干擾的情況；系統採用最新的壓縮技術mpeg4advancedaudiocoding （ aac ）和spectralbandreplication （ sbr ） ，音質足可與fm廣播媲美。
  6. Drm is a technology aims to improve sound quality of am transmission below 30 mhz and to reduce interference common in medium wave environment. new compression technologies known as mpeg4 advanced audio coding and spectral band replication are employed to offer sound quality comparable to that of fm transmission

    它是針對30mhz以下的調幅am廣播而設計，大大提高了音質，並改善了中波廣播較易受干擾的情況系統採用最新的壓縮技術mpeg4 advanced audio coding aac和spectral band replication sbr ，音質足可與fm廣播媲美。
  7. There ’ re two parts in the thesis : the design and implementation of fpga module and the design and implementation of the aes digital audio i / o circuit. i use xilinx corporation ’ s ise4. 2 as development tools to carry on fpga design, including hdl coding, functional simulation, logic synthesis, place & route and generation of programming files. fpga is used to implement audio routing, sine wave, adc ’ s calibration and led etc, . also, i have downloaded the configuration program into fpga chipset using mcu. eventually, the device is tested and the requirements of design is met

    通過測試， etheraudio音頻路由器完全達到了設計要求。 etheraudio音頻路由器完全符合aes / ebu硬體規范，滿足專業音頻傳輸、路由需求。最後，本文還介紹了etheraudio音頻路由器在廣播電臺中的應用實例，通過分析該音頻路由器在廣播電臺的應用方式說明本課題的實際應用價值。
  8. In this paper, we briefly introduced the performance of wave coding and vocoder, emphasizedly studied the principle and performance of variable rate vocoder q4401, including the internal construction and pins, qcelp coder & vocoder, pcm interface, cpu interface initialization process, command format and so on. we also designed a application circuit, with the experiment validated its performance. in this design, the pcm interface chip is tp3057, it was used to finish a / d transform, the compress coding was finished by q4401, the initialization and control were accomplished by 8051 singlechip

    重點是研究變速率語音編解碼晶元q4401的工作原理及性能。其中包括q4401的內部結構及管腳、 qcelp編碼方式、 pcm介面、 cpu介面、初始化過程、命令格式等，並在此基礎上，設計一個實際的應用電路，通過實驗，驗證其性能。在設計中用pcm介面晶元tp3057來完成從模擬信號到數字信號的轉換，再由q4401進行壓縮編碼，對q4401的初始化及控制由8051單片機來完成。
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