音頻解碼器 的英文怎麼說

中文拼音 [yīnbīnjiě]
音頻解碼器 英文
audio decoder
  • : 名詞1. (聲音) sound 2. (消息) news; tidings 3. [物理學] (音質) tone 4. (姓氏) a surname
  • : Ⅰ形容詞(次數多) frequent Ⅱ副詞(屢次) frequently; repeatedly Ⅲ名詞1 [物理學] (物體每秒鐘振動...
  • : 解動詞(解送) send under guard
  • : 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
  • 音頻 : [物理學] [電學] audio frequency; vf (voice frequency)音頻電路 voice frequency circuit; 音頻振蕩...
  • 解碼器 : codec
  • 解碼 : decoding; decipher; decode
  1. 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編的第16bit中的這個控制代指令告訴放在cd唱機中的hdcd,把訊號中原本的那個短促的陡高還原出來。這樣, hdcd就能夠在cd的44 . 1khz的取樣率極限內,把寬超過20khz的聲訊號重現出來了。
  2. Section ii describes the design approach and implementation of speech module on mcf5249 coldfire core. the speech codec optimizes g. 729a codes and added voice activity detection of g. 729b to save bandwidth ; the implementation of acoustic echo cancellation uses nlms algorithm and it can reduce echo though designing adaptive fir filter and speech detector ; the dtmf and cpt generate signal using two second order digital sinusoidal oscillators and detect signal by picking up the frequency information. but only get the frequency information is not enough in cpt detector, this thesis introduces a method

    其中對語的設計採用優化g . 729a代達到設計要求,並在此基礎上加入g . 729b的靜檢測模塊,以進一步降低網路傳輸帶寬;對回聲消除的設計採用nlms演算法,通過設計自適應fir濾波和語檢測達到回聲消除目的;對雙設計,信號發生端採用構造靜態參數表並通過二階正弦振蕩產生信號,信號檢測端提取率信息以檢測信號;對呼叫進程設計,除了類似雙的信號發生及率檢測設計外,還需要檢測信號持續時間,作者設計了一種基於匹配狀態表的方法以檢測信號持續時間。
  3. With argumentation and comparing the difficulty of developing in different ways, in multimedia communication the terminal make up with mpeg - 4 encoder and decoder based on vw2010 chip, main controller based on x86 core and pc compatible stpc soc chip, ethernet adapter based on rtl8139. and this hardware platform can run linux operation system. the following works are done by auther. 1. participate in designing the project of multimedia terminal

    經過論證和比較,權衡了開發難度和系統性能,在多媒體視會議終端設計中採用了基於x86內核片上系統晶元stpc的主控制模塊,基於vw2010晶元的mpeg - 4模塊,基於rtl8139晶元的10m / 100m以太網適配模塊的硬體架構,主控制模塊運行linux操作系統,作為多媒體終端的軟體運行環境及軟體開發環境。
  4. Our object is an intermediate frequency modem of a software defined radio transmitter - receiver of multi - service, multi - modulationmode and multi - processdatarate. first, related software defined radio theory is introduced ; later, channels of transmitter - receiver are designed with consideration of data format, modulation, fec, interwaving. and scrambling ; emphasis is placed on theory and implementation of an audio compression algorithm cvsd ( continuous variable slope delta modulation ) and a fec technique convolutional coding - decoding

    本文首先介紹了相關的軟體無線電理論;然後完成了包括數據格式,調制方式,糾錯方式,交織和擾等部分的中數據機通道設計;接著著重介紹了系統中使用的壓縮演算法cvsd (連續可變斜率增量調制)的原理和實現,以及作為前向糾錯的卷積理論和編的高效實現。
  5. Design of embedded audio system based on s3c2410 and uda

    1341型立體聲的嵌入式系統設計
  6. Specification for audio recording - pcm encoder decoder system

    記錄規范.脈沖編調制譯系統
  7. But large quantity of memory is often used and the area and power of the memory units usually hold the largest percentage of the whole chip

    片內存貯件過多導致存貯件的面積和功耗大大增加。該論文以ac3程序為例,提出了soc中存貯優化的一般方法。
  8. At last the audio - visual filter can get and regroup the pcm audio data which transformed into the time - frequency signals by the fft. at the same time the power spectrum can be displaying on the windows by using the win32 api functions. finally the test of this system shows the processes about transport stream of receiving in the collecting card, pushing in the bridge, separating in the parser, displaying in the audio - visual filter are satisfied with the data rata at the server

    效果filter內部通過獲取后的pcm格式數據,並利用緩沖管理對這些數據進行重新分組,然後對分組后的數據進行fft變換將時域信號變換到域之中,同時對變換后的數據作數學處理后得到其功率值,最後利用api畫圖函數對這些功率值進行了實時的特殊效果顯示。
  9. The main function modules discussed in this paper include : stream media protocols application model and realization, ts parsing module, audio / video decoder, audio / video synchronization model and realization, player memory buffer management module, multi _ task tech under uclinux. we also discuss the difference of the realization of stream media player between two defferent service types : broadcast tv ( btv ) and video - on - demand ( vod )

    從功能上,流媒體播放主要包含幾個大的功能模塊:流媒體協議棧的應用模型及實現機制、多節目復用傳輸流( ts )的析實現、媒體數據的同步機制的設計和實現方法、播放內存管理模型的設計和實現、 uclinux下多任務的實時調度和高效數據交互技術等。
  10. We are developing advanced ip core of system lsi for digital av application, such as a video codec and an audio codec, with high performance, high quality, small size and low power consumption

    我們正從事數字影產品先進ip核的開發,例如高性能、高質量、小尺寸和低功耗的視
  11. Per channel, and the result from ti ' s software simulator in ccs ( code composer studio ) is given, putting forward the principles and difficulties of realization of this algorithm. the recovered music signal is very close to the original signal and they are difficult to tell apart. in this paper a scheme of real - time implementation for this algorithm is discussed

    本文敘述了mpeglayer壓縮編的演算法及模擬實現研究,用c語言實現了的高保真樂信號64kbps每聲道的非實時,並在ti的ccs ( codecomposerstudio )系統中的軟體模擬上進行實時研究,提出了該演算法在具體實現中的要點和難點。
  12. 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cdma通信系統的變速率語標準,它實現了語的多種低速編,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,可以在保證高質量語的同時盡可能增加系統的容量,被認為是變速率語在cdma系統中應用的「突破性」技術,代表了當前語發展的方向和潮流,因此smv的研究具有很大的價值。
  13. On the basis of analyzing the old system and theory, the element circuits of wireless digital audio transceiver modules are designed in detail including the digital audio encoding and decoding circuits with the surrounding circuits, the fsk circuit based on pll frequency synthesizer, the power amplifier circuit, the frequency discrimination and agc circuit

    在分析原系統結構和理論的基礎上,完成了整個無線數字傳輸模塊各單元電路的設計。主要包括有數字電路及外圍電路的設計、基於鎖相率合成理論的fsk電路設計、功率放大的設計、鑒與agc控制電路的設計。
  14. This dissertation is concentrated on the design of ess decode project, which make use of mpeg decode ic ess6425 as the core supported by other ic cs4340 ( audio ic ), gl811 ( usb 2. 0 to ata / atapi bridge controller ), etc. on the basis of the decode project, a multi - media mp4 hd player is designed with special features of convenient installment, competitive price and perfect performance

    本研究以美國ess公司的mpeg專用晶元es6425為核心整合其他輔助晶元如cs4340 (處理晶元) , gl81 ( 1usb2 . 0toata / atapibridgecontroller )等,設計出一款mpeg - 4多媒體播放,該播放具有安裝簡單、價格低廉,功能全面等特點。
  15. It includes the mepg2 demodulation module, the numeral regards the audio frequency coder and the conditional access module and so on

    它主要包括mepg2調模塊、數字視和條件接收模塊等。
  16. This paper also provides some background knowledge about sip and video conference, such as the sip framework, the history of voip and video conference, etc. this paper emphasizes on the implementation of sip terminal on the windows platform, including the realization of sip stack, the capture and compression of video and audio steams, rtp transmission of stream data, the working thread of sip terminal and the optimization of video codecs

    本文在論述研究工作的同時也介紹了一些相關的背景知識如sip協議的框架、 voip和可視電話的發展歷史等。本文重點論述了windows平臺sip多媒體終端的實現,包括sip協議棧的實現、視流的採集、編和網路傳輸方案、 sip終端的工作流程、以及編基於指令集的優化。
  17. Second, this dissertation implements separately a mpeg - 2 video decoder and a dolby ac - 3 digital audio decoder based on software mode, and gives a audio & video synchronization algorithm based on audio - clock - benchmark in mpeg - 2 system decoder, whose feasibility and practicability have been proved by experimenting. it is an all - purpose algorithm, which can perform different decoder according to mpeg - 1 or mpeg - 2 system models, and can also be used for reference to the implementation of other multiplex stream decoders

    然後,論文實現了基於軟體方式的mpeg - 2標準視及ac - 3格式壓縮的實時與回放,並依據mpeg - 2系統模型實現了一種基於基準時鐘的mpeg的視同步演算法,實驗證明該演算法可行、實用、通用性好,對符合mpeg - 1或mpeg - 2系統標準的視音頻解碼器均具適用性。
  18. Thirdly, this dissertation gives a synchronization algorithm of double work - stations " playing out based on digital reference information

    同時,該演算法也對mpeg硬體及其他多路復用格式的視壓具有借鑒價值。
  19. Install audio decode tone exceed convulse

    內置音頻解碼器質超震撼
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