audio bit 中文意思是什麼

audio bit 解釋
音頻信號位
  • audio : adj. 【無線電】聽覺的，聲音的，音頻的；成音的。
  • bit : n 1 少許，一點兒，一些；（食物的）一口，少量食物。 〈pl 〉 吃剩的食物；小片。2 〈口語〉一會兒，一...
  1. Television - 24 - bit digital audio format for hdtv bit - serial interface

    電視. hdtv位串列介面用24位數字音頻格式
  2. Resemble with the digital watermark methods of the other media such as still image and video, digital audio watermark must account for some problems which consist of perceptual transparency, data bit rate, robustness, security and real - time etc. the robustness of watermark is vital to the practical application, which requires the watermark providing with significant data since it suffered from some intended attacks or unintended revisal

    與靜態圖像、視頻等數字水印方法類似，音頻水印的研究主要須解決感知透明性、數據嵌入率、魯棒性、安全性以及實時性等問題。水印演算法的魯棒性對于實際應用來說是至關重要的，它要求數字水印在遭受有意的攻擊或無意的修改後，仍能提供有意義的數據，這一性能對版權保護的應用顯得尤為突出。
  3. 24 - bit digital audio format as ancillary data signals in hdtv serial interfaces

    高清晰度電視串列介面中作為附屬數據信號的24比特數字音頻格式
  4. The 32 - bit cpu core with enhanced multiply accumulate emac unit provides optimum performance and code density for the combination of control code and signal processing required for mp3 decode, file management, and system control. fs2401clqn is a single - chip mp3 audio decoder

    當用戶端的pc與存儲媒介之間透過usb 2 . 0介面做資料交換時，因僅需要cpu最低程度的參與，而大多是以硬體處理方式，所以可以達到高速傳輸的目的。
  5. Study on algorithms of low bit rate hi - fi audio coding based on wavelet transform and psychoacoustic model

    基於小波變換和音質模型的音頻編碼演算法研究
  6. An algorithm of fec ( forward error correction ) system and it ’ s fpga implementation were researched, . then the fec system was applied to an audio transmission platform. the platform can decrease the error bit rate to 1 10 - 7 when channel error bit rate is 3 10 - 3. veriloghdl was chosen to design the circuits

    研究了一種前向糾錯（ fec ）演算法及基於fpga的相應電路設計，將此電路應用於數字音頻無線傳輸，搭建了一個完整的數字音頻無線傳輸平臺，當無線通道誤碼率為3 10 - 3時，經過該糾錯電路可以降低到1 10 - 7以下。
  7. The project uses for reference the algorithm thought of sbc ( subband coding ) to measure off the audio to the corresponding frequency width and encode it by the different sensitivity of human hearing, which results in the lower coding rate and bearable voice quality. the algorithm processing low bit - rate audio is designed to be self - adaptive by the situation of network. the component developped by that algorithm and project has already been used in the realtime interactive educational system

    該方案借鑒sbc （ subbandcoding ）子帶編碼演算法思想，將音頻按對人聽覺敏感程度不同劃分為相應的頻帶並進行相應的編碼，從而得到較低的編碼率和較好的語音質量，設計了可根據網路狀況進行自適應的低帶寬音頻處理演算法。
  8. Underlying audio format for the internet does not allow the use of 24 - bit data, and the sound card of the reader would have to be able to render 4g. the operating principle should be crystal now

    不過要注意的是， 24 - bit格式的資料並無法在網際網路上應用，你的聲卡必須具備支援4g的能力才能使用，這樣子你明白了吧。
  9. 2. when the existed dynamic bit rate allocation algorithms is working, the video data bit rate and the audio data bit rate are changed

    2 ）已有的動態速率調整演算法在進行動態速率調整時，音、視頻的位速率都要改變。
  10. Telecommunications - digital processing of program audio signals - algorithm for 15 - khz audio at 384 kbit s using 14 11 bit coding

    電信.聲頻信號程序的數字處理.使用14 11位編碼以384k位秒速率傳輸15k赫茲聲頻的演算法
  11. Parametric bit allocation technique in audio compression coding

    音頻壓縮編碼中的參數比特分配技術
  12. H. 323 is the standard about multimedia communication released by itu - t. tm1300 including a very powerful, general - purpose vliw processor core ( the dspcpu ) that coordinates all on - chip activities is a media processor for high - performance multimedia applications that deal with high - quality video and audio. the dspcpu implements a 32 - bit linear address space and 128, fully general - purpose 32 - bit registers

    H . 323是itu ? t推出的用於ip分組網路的多媒體通信終端協議， trimediatm1300處理器晶元是philips公司推出的一種基於多媒體應用的具有vliw指令，含有128個通用寄存器， 32位的高性能處理器，它能夠通過編程實現通信協議，完成高質量的音頻、視頻處理和網路介面。
  13. No line matching interface clip supporting format pcm _ signed, 11025. 0 hz, 16 bit, mono, little - endian, audio data, and buffers of 16760 to 16760 bytes is supported

    使用虛擬機運行很正常，可是放到手機里邊就不行啦，求各位高人相助
  14. Furthermore, on the analysis of the psychoacoustic models, some simplification can be made in the psychoacoustic models. the purpose of this simplification is to reduce the bit rate of audio stream

    此外，基於對心理聲學模型的分析，對心理聲學模型作若干近似，簡化的目的是為了用較少的比特編碼音頻數據。
  15. At first the article puts emphasis on analyzing those current network ip technology, various audio codec algorithms, realtime stream medium transmit technology and those process mechanism of realtime low bandwidth audio stream medium, etc. in allusion to high requirement of system, resulted from so many terminations of attending a lecture, rate of flow, bad situation of network and realtime interactive voice, a new algorithm and the relevant project of processing low bit - rate audio stream was brought forward

    本文著重分析了當前網路方面的ip技術，各種音頻編碼演算法，實時流媒體傳輸技術，實時低帶寬音頻流媒體的處理機制等等。針對網路實時應用中諸如客戶端眾多，各種多媒體數據流量大，網路狀況差，語音交互實時性等方面較高的要求，提出一種新的實時低帶寬音頻流處理演算法及相應的處理方案。
  16. Recently, advances in postprocessing mechanisms have been studied to improve lip synchronization of head - and - shoulder video coding at a very low bit rate by using the knowledge of decoded audio in order to correct the positions of the lips of the speaker [ 3. 36 ], figure 3. 2 shows an example of the block diagram of such a postprocessing operation

    最近，對改善在很低比特率時頭肩像視頻編碼的唇同步問題的后處理機制的研究已經取得進展，這種機制運用解碼音頻的知識校正講話者的唇位，圖3 . 2顯示了一例這類后處理過程的框圖。
  17. The iso / mpeg - audio layer iii digital audio compression scheme is the most powerful of the three audio compression algorithms standardized in iso / mpeg dis - 111 72. even at very low bit - rate of only 64kb / s, layer iii is one of the most excellent compression tools for applications such as real - time network radio and high fidelity music. this paper describes a study on algorithm simulation of mpeg / audio compression layer iii

    Mpeg / audiolayer提供了iso / mpegdis - 11172標準的三個音頻壓縮方案中最強的壓縮能力，即使工作於64kbps ，仍能保證高品質的音響效果，是需要以低碼率傳送數字高保真音頻信號的應用領域的理想工具。
  18. Comparing with the existed dynamic bit rate allocation algorithms, the model can provide better audio data quality by adjusting the video data bit rate. this meets the requirement in remote education system

    本模塊只對視頻位速率進行調整，保持音頻位速率不變，保證音頻質量，在一定程度能滿足遠程教育的需要。
  19. Based on the research of videoconference systems of h. 323 protocol over ip networks and the author ' s experiences of implementing h. 323 videoconference systems in remote education area, in this thesis the main factors that affect videoconference quality are analyzed, and a dynamic bit rate allocation model is proposed and partly implemented. this model is designed to dynamically allocate bit rate for multi - media data flow ( including audio data and video data ) in fixed bandwidth network environment. when continuous multi - media packet losses are detected in ip based h. 323 videoconference system, the bit rate of video data is adjusted meanwhile the bit rate of audio data remains unchanged, and the bit rate allocation of multi - media data ( including audio data and video data ) is optimized as a whole effect

    本文結合作者在h . 323視頻會議系統應用於遠程教育的經驗，通過對現有的基於ip網路的h . 323協議視頻會議系統的研究，分析了影響視頻會議質量的原因，提出並部分實現了在固定帶寬的網路環境下，基於ip的h . 323視頻會議的多媒體數據包發生丟包、抖動或延時時，保持音頻數據位速率不變，通過對視頻數據的位速率的進行調整，最終實現旨在提高視頻會議語音質量的多媒體數據位速率動態調整的模型。
  20. Mke - 810 encoder is able to provide high - quality video, audio compression with mpeg - 2 encoder. it provides efficient rate and cache control can be both high and low bit - rate high - definition quality. it mp ml with mpeg - 2 video coding standard resolution of 720 576 25. 2m - 20m programs for each output bit rate adjustable. meanwhile a variety of interface and high performance core of the larger independent mpeg - 2 module integrated within the box. meet efficient installation

    採用4 : 2 : 2數字視頻和20位數字音頻編解碼技術內置時基校正和幀同步功能，保證信號的傳輸質量單模傳輸方式，波長為1310nm或1550nm ，傳輸距離可達100公里
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