語音編碼器 的英文怎麼說

中文拼音 [yīnbiān]
語音編碼器 英文
vco voice coder
  • : 語動詞[書面語] (告訴) tell; inform
  • : 名詞1. (聲音) sound 2. (消息) news; tidings 3. [物理學] (音質) tone 4. (姓氏) a surname
  • : Ⅰ動詞1 (編織) weave; plait; braid 2 (組織; 排列) make a list; arrange in a list; organize; gr...
  • : 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...
  • : 名詞1. (器具) implement; utensil; ware 2. (器官) organ 3. (度量; 才能) capacity; talent 4. (姓氏) a surname
  • 語音 : speech sounds; pronunciation; voice
  • 編碼器 : (將一項信息變換成一系列數碼信號的電路) coder; encoder; encipheror編碼器方框圖 encoder block diagram
  • 編碼 : encoded; code; coded; encrypt; codogram; coding編碼表 encode table; 編碼程序 builder; 編碼尺 code...
  1. 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

    本文結合當前技術尤其是混合技術的驚人發展及數字信號處理的日益實用化,研究了技術,並重點討論了變速率技術。論文簡要介紹技術中的波形和聲的主要性能。
  2. The optimal gain filter of ld - celp

    低延遲激勵演算法的最佳增益濾波
  3. With the developing of vlsi in recent years, high function dsp has been produced ( such as tms320 series dsp produced by ti ) and their cost is dropping. thus, this established the foundation for making complex speech coder practical and producible. the paper researched and discussed the fix - point real implementation of g. 728 by dsp tms320c5402 chip

    但是,近幾年來,隨著大規模集成電路( vlsi )的發展,已生產出高性能數字信號處理晶元(例如ti的tms320系列dsp晶元) ,而且其成本在不斷降低,這就為復雜的語音編碼器的實用化和產品化奠定了基礎。
  4. At the present time, evrc is the best vocoder in the cdma system when take into account both the voice quality and the encode rate

    在目前的cdma系統中,綜合質量和速率, evrc是最佳的語音編碼器
  5. 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數字信號進一步加強。
  6. We select fpga of type xc3s200 as hardware to design the coder and display the hardware resources inside, moreover study the method and steps of designing dsp, based on fpga, by using system generator, finally, it emphasizes the design process of multi - band excitation vocoder. we can work out the module of high pass filter and the module of low pass filter, module of divide frame, module of keynote rough estimate, module of keynote fine estimate, module of band - separated v / u judgment / verdict and module of band - separated amplitude estimate, by using simulink, ise and system generator

    本文選用型號為xc3s200的fpga作為設計的核心硬體,介紹了其內部所含的硬體資源,並研究了利用systemgenerator基於fpga設計dsp的方法和步驟,最後,本文把重點放在多帶激勵語音編碼器的設計上,利用simulink , ise和systemgenerator分別設計其中的高通低通濾波模塊、分幀疊加模塊、基粗估模塊、基精細估計模塊、分帶v / u判決模塊、分帶幅度估計模塊。
  7. The evrc vocoder is varialbe rate, the maximal encode rate is skbps, its voice quality is closed to qcelp - 13k, and has better ability of anti - disturbance

    Evrc聲是可變速率的,最大速率為8kbps ,在話質量上接近於qcelp - 13k的語音編碼器,且具有更好的抗干擾能力。
  8. Fpga has become the best selection in the design of complex digital system in modern design process of digital system, especially in communication system, because of its merits like high integration, better reliability, short design period, less investment and agility. the usage efficiency of fpga for communication system can be raised by designing low rate speech coder in fpga

    在現代數字系統設計中, fpga因為高集成度,高可靠性,設計周期短和投資小逐步成為復雜數字系統設計的理想首選,尤其是在通信系統中大量地使用,把低速率的語音編碼器在fpga中設計,可以提高通信系統中的fpga的利用率,節約成本。
  9. The neural network for gain filter in speech code algorithm

    演算法的神經網路增益濾波
  10. In this paper, we give a detail discussion on the key technology, including software and hardware designing of g. 729a multi - channel speech codec realtime implemention on a simple dsp processor - tms320c6202. in combination with the requirements of a military communication network, the atm adaptation solution of g. 729 coder bit stream is analyzed. a kind of new atm adaptation technology - aal2 is introduced. the analyse and research of aal2 are provided

    本文詳細討論了多路g . 729a在一片dsp處理tms320c6202上實時實現的軟硬體設計和關鍵技術。結合某軍事通信網設備的需要,進而對g . 729流的atm適配方案進行了分析。提出了用一種新的atm適配技術- - aal2進行適配的方案。
  11. Firstly, it introduces the development of speech coding, along with the significance of the low bit rate speech coding. it also compares the model of traditional dualistic excitation lpc vocoder and the multi - band excitation vocoder, and lucubrates the analytical method of frequency domain and time domain in the parameter extraction of multi - band excitation vocoding. secondly, based on the parameter extraction operation of keynote cycle, it adopts time domain in rough estimate operation of keynote and frequency domain in fine estimate operation of keynote, in according to the immediacy required in practice, to minish operation amount

    本文闡述了一種基於fpga的多帶激勵語音編碼器的研究與設計,首先介紹研究的發展狀況以及低速率研究的意義,接著對比分析了傳統二元激勵lpc聲模型和多帶激勵模型,並深入研究了多帶激勵參數提取的頻域和時域分析法,然後根據實際應用的實時性要求,為了減小運算量,在基周期參數的提取的演算法實現上,本文採用在時域進行基粗估運算,在頻域進行基精細估計運算。
  12. Tdma third generation wireless - adaptive multi - rate speech codec minimum performance requirements

    時分多址聯接第三代無線電通信.自適應多速率最低性能要求
  13. The optimized g. 729 speech codec is tested by all the itu test sequences and has accomplished the request

    用itu - t提供的g 729語音編碼器測試序列進行測試,基本上達到了要求。
  14. Finally, discuss the design and realization of tcp / ip network based on the vehicle - carried communication system in details. furthermore, depict the 13 - bit linear pcm code - filter about motorola 145483 chip, which is used in the system

    最後,對車載通訊系統的網路設計方案以及實現方法進行了詳細介紹,並對13位的pcm語音編碼器(摩托羅拉mc145483晶元)進行了介紹。
  15. The tcp / ip network and the driver of voice coder are mainly introduced. the basic content is as follows : firstly, plainly introduce the development of embedded system and the outlook for embedded system, moreover, consider the application of the embedded developing tool

    其中對tcp ip網路的設計和語音編碼器驅動的實現進行了詳細介紹,其基本內容如下:首先,簡單介紹了嵌入式發展的過程及其前景,同時也介紹了嵌入式開發工具的具體應用。
  16. In the next, we discuss the system of the meg - 1 layer i. the paper centers on the two kernel sub - parts : filtering coding and psychoacoustic model, do some research work in sub - band coding ( cbc ) theory and the relate theory such as quadrature mirror filter ( qmf ) and analyse sub - band filter ; also do research work in psychoacoustic theory especially the part related to the mpeg - 1 layer i. in the third chapter, introduce the ti tms320c6000 series dsps and their characteristics, also about the software development flow and the ti dsp / bios operating system of it. the forth chapter is the most important, firstly, according the algorithm flow in protocol, using c language validate the algorithm ; then, transplant and optimize the coding in dsp. in the processing of optimize, acording the assembler program characteristic of ti dsp, the paper put forward the analyse sub - band filter dsp optimization algorithm base on the eight spot idct. the algorithm has been optimize have greatly improved the work efficiency. make use of the technology of the dsp / bios host channels, data io pipe, software interrupt, we implement the musicam algorithm base on dsp / bios

    論文首先對當前技術的發展、分類以及mpeg系列頻標準作了介紹;接著在第二章,給出了layer的musicam ( masking - patternuniversalsubbandintegratedcodingandmultiplexing )演算法的系統組成,圍繞分析子帶濾波和心理聲學模型兩個核心模塊,深入研究了子帶工作原理、比特分配及子帶中用到的正交鏡像濾波和分析子帶濾波;探討了心理聲學基本原理和mpeg . 1layer所用到的心理聲學模型。第三章對titms320c6000系列dsp作了簡介,介紹了6000系列dsp結構特點、 c6000dsp軟體開發流程和tidsp / bios操作系統。第四章是本文的重點,首先根據協議給出的演算法用標準c程實現並調試通過。
  17. It adapts to the cdma system and achieves multi - rate speech coding and decoding. source and mode control are combines in smv for rate selection, so it improves the flexibility of cdma system, it will allow cdma subscribers to enjoy superior quality while allowing service providers to increase capacity as needed. smv is regarded as a breakthrough technology that provides significant capacity and quality gains on cdma systems, so the researching of smv is of great practical value

    可選模式聲( smv ? selectablemodevocoder )是3gpp2最新的用於寬帶擴頻cdma通信系統的變速率標準,它實現了的多種低速和解,在速率選擇上將源控和模式控制相結合,提高了cdma系統的靈活性,可以在保證高質量的同時盡可能增加系統的容量,被認為是變速率在cdma系統中應用的「突破性」技術,代表了當前發展的方向和潮流,因此smv的研究具有很大的價值。
  18. Pitch detection is one of the most important tasks in low - rate speech coding field. the accuracy of pitch detection will affect the performance of the whole codec

    檢測是低速率領域的一個非常重要的問題,準確檢測信號的基周期非常關鍵,它直接影響到整個聲的性能。
  19. At first, from the discrete digital model of the speech generation, the basic of speech encoding technology is briefly introduced in this thesis. secondly, some key techniques in the linear predictive speech coder of g. 723. 1 arithmetic are discussed in detail

    本文首先從產生的離散數字模型出發,簡要敘述了的技術基礎,詳細討論了g . 723 . 1標準的線性預測的一些關鍵技術。
  20. In section two, we introduce some of the main coding theories that was in the g. 729, including the digital model of the production of speech signal, the linear prediction coding, vector quantization, lbg arithmetic and perceptual weighted filtering

    第二章著重討論了g 729壓縮協議中所涉及到的基礎理論。主要包括信號產生的數字模型、信號的線性預測分析、矢量量化及其lbg演算法和感知加權濾波
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