signal compression 中文意思是什麼

signal compression 解釋
信號壓縮
  • signal : n 1 信號，暗號；信號器。2 動機，導火線 （for）。3 預兆，徵象。adj 1 暗號的，作信號用的。2 顯著的...
  • compression : n. 1. 壓縮；壓緊；濃縮，緊縮。2. 加壓；壓抑。3. （表現的）簡練。4. 應壓試驗。
  1. 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。
  2. Based on the theory of pulse compression, the essentiality of lfm signal is discussed

    首先介紹了線性調頻信號脈沖壓縮的實現方法。
  3. 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。
  4. With the established rf front - end system simulation platform, adding the digital modulated baseband signal, this paper simulated the multifold digital modulated signal ’ s transmission, such as 2ask, qpsk, and 16qam. then researches of power compression and phase noise of local oscillation influence the bit error ratio for different modulated system. the designing is satisfied multifold functions request with the high - powered and integrated broadband rf front - end

    隨后在建立的寬帶射頻前端通用模擬平臺上，加入基帶數字調制信號，對多種數字調制格式的信號在該通用平臺上的傳輸作了研究，模擬了2ask 、 qpsk和16qam等調制格式信號的發射與接收，研究了功率壓縮和本振相位噪聲對不同調制的誤碼率影響，實現了滿足多種功能要求的寬帶高性能綜合射頻前端的設計。
  5. Experimental result shows that for sonant part of speech signal, 3 ~ 5 common ridges is enough to describe the main characteristics. signal compression is achieved by choosing proper way to represent the ridge information and use it to reconstruct the original signal

    在信號重建過程中，選擇合適的方法用少量數據來描述起關鍵作用的參數，並用這些參數來重建信號，可以達到信號壓縮的目的。
  6. 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。
  7. The experiment result shows that the coding and decoding rates are high, and the signal to noise ratio, compression rate and visual quality are quite good

    實驗結果表明該方法編（解）碼速度快，並有較好的信噪比、壓縮比及視覺效果。
  8. The scrambling and descrambling technologies include non - disturbing frequency method, analog base band disturbing method, the radio - frequency signal disturbing method, the digital disturbing method for the analog signal and the digital signal disturbing method, etc. after the introduction a project based on the compression of the rf synchronous information is formulated

    本文綜述了目前常用的幾種主要的電視信號的加解擾技術，如非擾頻加解擾技術、模擬基帶加解擾技術、射頻信號加解擾技術、模擬信號的數字處理加解擾技術以及數字信號加解擾技術等。
  9. Experimental results show that the embedded watermark is robust against various signal processing and compression attacks

    實驗證明經過信號處理與壓縮后，水印仍然可以被檢測出來。
  10. Arithmetic figure that new dvd that conduct and actions world the adoption arithmetic figure of the first style see the port dvi port to broadcast, three stars the dvd - hd938 to can output to have not yet the compression to see the signal, but have no to need to pass by few molds the conversion process, from but avoided the loss of the painting quality with lose true, deliver the real arithmetic figure signal to television the dvi port is a high take the ieee1394 port breadth port, its deliver the speed to attain 5 gbps, is more than ten times to deliver the speed. at the same time, need not proceeding of port of adoption dvi few mold conversion, can avoid to convert what process result in to lose true, from but real realizes the arithmetic figure deliver

    作為世界第一款採用數字視頻埠dvi埠的新型dvd播放器，三星dvd - hd938能夠輸出未經壓縮的數字視頻信號，而無需經過數模轉換過程，從而避免了畫質的損失和失真，向電視傳輸真正的數字信號dvi埠是一種高帶寬的埠，其傳輸速度達到5gbps ，是ieee1394埠傳輸速度的十幾倍。同時，採用dvi埠不用進行數模轉換，可以避免轉化過程造成的失真，從而真正實現了數字傳輸。
  11. At the receiving end, a inverse process is performed. the system receives low rate data and the fpga reorganizes a frame of data which is decoded by the compression chip every 20 ms. the obtained pcm signal is performed d / a to restore the analog speech signal

    在收端進行相反的過程，接收低碼率數據，並由fpga重新組幀，送至主晶元解碼得到pcm信號，再作d / a變換，恢復出模擬語音，系統是全雙工的。
  12. The primary advances in speech and audio signal processing that contributed to multimedia applications are in the areas of speech and audio signal compression, speech synthesis, acoustic processing, echo control and network echo cancellation

    語音和音頻信號處理的改進對多媒體應用的貢獻在下述范圍：語音和音頻信號壓縮、語音合成、聲學處理、回聲控制以及網路回聲消除。
  13. 2. analyse lfm signal compression performance with high compression ratio ， program the correlative test software ， compare the pulse compression performance with various window - functions and get the satisfied results. 3

    分析了lfm信號在高壓縮比情況下的壓縮性能，並編制了相關測試軟體，對不同加窗情況下的脈沖壓縮性能進行了比較，取得了滿意的效果。
  14. Then, we give tile introduction of the theory of digital audio signal compression, the algorithm and somp accelerate algorithm for the mp3 encode and decode. ln the end, we propose a parallel processing system with two dsp to be a real - time mp3 encoder and decoder

    然後，詳細介紹了數字音頻信號壓縮的原理， mp3編碼和解碼的演算法實現，並採用了一些編碼和解碼的加速演算法，最後提出了一個雙dsp并行處理系統用於實時實現mp3編碼和解碼。
  15. We study the wavelet neural network theory and how to build models. the wavelet networks can be used in ecg signal compression by adjusting wavelet basis and weight values. at the same time, this algorithm also can reconstruct the ecg signal very well

    在理論研究了小波變換方法和神經網路的基礎上，提出了基於小波神經網路的ecg數據壓縮演算法，並分析研究了基於小波神經網路壓縮ecg數據的原理和模型的構建方法。
  16. A flexible and efficient sinusoidal modeling using matching pursuits suited for signal compression

    一種適于信號壓縮正弦模型的匹配跟蹤方法
  17. Still, we generalize our work and give a preview of the prospect in the digital audio signal compression field

    在論文的最後，對本文所做的工作做了總結並展望了該領域的前景。
  18. Because of enormous volume of video digital signal, this requires more advanced technique in video signal compression

    由於視頻圖像信號具有龐大數據量，這就對視頻圖像壓縮技術提出很高的要求。
  19. In order to unfold the whole essay, a general description of the presellt research condition in the field of digital audio signal processing especially in the field of digital audio signal compression is presented

    本文針對當前實時實現mp3音頻編碼和解碼的研究課題，首先概述了當前數字音頻信號領域尤其是音頻信號壓縮領域的研究現狀。
  20. With the rapidly development of multimedia signal compression and broadband networks technologies, methods of transmitting high - quality multimedia information via networks have become one hottest point of it technologies

    隨著多媒體信息壓縮技術和寬帶網路技術的快速發展，通過網路傳輸高質量的多媒體信息已經成為信息技術領域研究的熱點之一。
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