音頻模數 的英文怎麼說

中文拼音 [yīnbīnmóshǔ]
音頻模數 英文
audio ad-da convertor
  • 音 : 名詞1. （聲音） sound 2. （消息） news; tidings 3. [物理學] （音質） tone 4. （姓氏） a surname
  • 頻 : 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...
  • 模 : 模名詞1. （模子） mould; pattern; matrix 2. （姓氏） a surname
  • 數 : 數副詞（屢次） frequently; repeatedly
  • 音頻 : [物理學] [電學] audio frequency; vf （voice frequency）音頻電路 voice frequency circuit; 音頻振蕩...
  • 模數 : [物理學] modulus; module; modulo; mod
  1. They use industry - standard sound system design software like leap - 5, lms and clio while our laboratory utilizes smaart test hardware and software. we constantly update and upgrade ourselves with the latest software and diagnostic technology in these fields. lax reassures you that our products are of the highest quality found in the world of professional audio merchandise

    其享譽全業的系統數字音頻產品，以全面系統專業的推廣模式，新穎強大的功能，簡便的操作和傑出的電聲水準，獲得消費者廣泛的好評，並逐步成為未來專業音響發展的主流。
  2. 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。
  3. Firstly, we study the construction of emotion - speech template database, and analyze the common features such as pitch, energy and formant. after choosing the useful features by using fuzzy entropy effectiveness analysis, we get better performance with the application of neural network. in addition, we propose some more efficient features such as speech rate, pitch slope, mel - frequency cepstral coefficients and its transient parameters, and design a processing model based on vector quantization for cepstral features to fusing different features

    本文首先介紹了情感語音數據庫的建立情況，然後研究了基音頻率、振幅能量和共振峰等目前常用的情感特徵在語音情感識別中的作用，並且通過一種基於模糊熵的特徵有效性分析方法進行了有效特徵的篩選，應用人工神經網路建立了初步的語音情感識別模型，經過實驗發現特徵篩選后系統的識別效果有著一定程度的提高。
  4. 192 khzs 24s bit few mold conversion

    192khz 24bit音頻數模轉換器dac
  5. The focus is placed on the investigation of the standard of the encoding algorithm for mpeg audio layer iii, and the analysis of the major four modules in the compression algorithm, including encoding of subband filter bank, psychoacoustics model, quantification and huffman coding, frame packing

    重點研究了mpeg音頻第層編碼的演算法標準。詳細分析了壓縮演算法中的四個主要功能模塊：子帶濾波器組編碼，心理聲學模型，比特流量化與霍夫曼編碼，幀數據流格式化。
  6. Main ideas as follows : research of audio digital watermark algorithm based on fast fourier transforms and psychoacoustic auditory model

    主要工作有:基於快速傅立葉變換與聲學模型的音頻數字水印演算法研究。
  7. Usually point a kind of inside to pack the arithmetic figure cent the network with the box of the power enlarger. arithmetic figure type the power enlarger for method for signal for inputting for arithmetic figure comparing, at with arithmetic figure signal handling again and again partitioning the empress, then and respectively these signals transformation is imitating the signal, then again from eachly from of box enlarge the empress to go to again to push the the cowgirl in the box to pronounce the unit

    通常指一種內裝數字分頻網路和功率放大器的音箱。數字式音箱輸入的信號為數字比特流，在用數字信號處理的方法將音頻頻譜分割后，便分別將這些信號變換為模擬信號，然後再由各自的功率放大器放大后再去推動音箱中的相應發音單元。
  8. The frequency analysis of the psychoacoustic model can give the inaudible, redundan elements of audio. in this paper we employ it to guarantee the inaudibility of the embedded watermark

    心理聲學模型的時頻域分析主要用於音頻數據的壓縮編碼，這里主要考慮將其用於音頻水印演算法中去，以便從理論的角度來考慮嵌入水印的不可感知性。
  9. Secondly, introduce discrete multi - tone modulation principle in detail, have a systemic and comprehensive analysis and explanation on minimum mean square error ( mmse ) channel shorten time domain equalizer design methods which are based on all kinds of cost functions, analyze their advantage and disadvantage. research on time domain equalizer structure, compare all the time domain equalization algorithm with simulation which afford a valuable reference for the choice of equalization algorithm and equalizer structure when design time domain equalizer

    詳細介紹離散多音頻調制原理，對mmse通道縮短法和基於其他代價函數的時域均衡器設計方法的進行了系統全面的分析和闡述，分析了各自的優缺點；對時域均衡器結構進行研究；模擬比較了各種時域均衡方法，為進行時域均衡器設計時均衡演算法、均衡器結構的選擇提供了有價值的參考。
  10. It realizes the functions of storing and transferring the information of fire alarm, displaying time, scanning keyboards, the safe maintenance of system and so on. the part of wireless communication is controlled by at89c2051 mcu, mainly realizing transmitting and receiving data between controller and other detectors. the external signals received by a wireless module are transmitted to mcu through shaping circuit, and the signals of mcu are transmitted by a wireless module through tone modulation circuit

    人機交互部分以單片機c8051f020為核心，為用戶提供一個良好的操作環境，實現了火災報警信息的存儲及調用、時間的顯示、鍵盤的掃描、系統的安全維護等功能;無線通信部分由單片機at89c2051來控制，主要實現控制器與其它探測裝置之間信號的無線發射和接收，無線通訊模塊接收到的外部信號經過整形電路送入單片機，單片機發出的信號經過音頻調制再由無線通訊模塊發送出去，這樣實現了數據的無線傳輸;本文還從節能的角度出發，兼顧性能的可靠性，提出了一種合理的無線火災報警的信息傳輸模式。
  11. With the brand - new concept of network copyright and leading model of digital frequency entertainment, “ listenworld ” initially steps on to the new phase of “ ear business ” in on inter - net business and provides customers with a brand - new model of receiving information

    「聽世界」網站以全新的網路正版概念和超前的音頻數字娛樂模型，搶先開創國內網際網路行業的「耳朵經濟」新局面，為用戶提供了一種全新的信息接收方式。
  12. This thesis presents a new audio digital watermark algorithm based on has model. the steps : first, it classifies the signals through fft ; second, calculating the tones, noise and the overall masking threshold of different phases through has, and changing the energy value of tones which are more than that of the overall masking threshold to embed the watermark information

    提出了一種基於人類聽覺系統（ has ）模型的音頻數字水印演算法，該演算法首先對信號進行分段離散快速傅立葉變換（ fft ） ，再根據人類聽覺系統模型計算出各段的類純音、類噪音以及各段的總體掩蔽閾值，通過改變大於總體遮蔽閾值的類純音的能量值來嵌入水印信息。
  13. The structure and circuits of high resolution adc ’ s modulator which applied in audio signal range with 20khz base band and 16 bits resolution have been researched

    本文對一種適用於音頻信號范圍的高精度模/數轉換器的調制器部分進行了結構和電路研究，基帶頻率20khz ，精度16位。
  14. The emergence of digital television is one of the most important things in the history of telecast. it can not only improve the quality of audio and video but also change the traditional mode of watching tv. it can provide many incremental operations, such as television website, video - on - demand, remote education and information services

    數字電視的出現是電視廣播史上最重大的事件之一，它不僅大大提高了電視音頻和視頻的質量，還改變了傳統的收視模式，能夠提供電視網站、視頻點播、遠程教育、信息服務等多項增值業務。
  15. First an analog video signal is decoded by saa7 11 a to form a digital video signal complying with ccir6o1 which then is compressed by an special chip ibms42o, at last, the video es is packed with audio es to form ts by computer

    具體來說，是把模擬視頻源解碼成符合ccir601規范要求的數字視頻源，經專用的mpeg2編碼晶元壓縮形成視頻es流送入計算機，與一路音頻es流打包復用后形成一路ts流。
  16. Dsp digital shift frequency sires is new generation conference system and teaching equipment, which adopts dsp technology. it can lock the point of howl a automatically, dsp digital shift frequency and aptitude mix sound of transmitter

    數字自動濾波器，數字話筒智能混音器，數字高低音調節器等多項數字化音頻處理模塊的技術優勢，解決了會議教學擴聲中的聲反饋問題，可提升話筒靈敏度
  17. A scheme to identify the stochastic signal and its modes by computer is introduced and using the method of the autocorrelation functions for recognizing the stochastic audio signal fleet has been applied

    摘要分析了利用自相關函數法實現快速識別隨機音頻信號，介紹一種用計算機實現隨機音頻信號處理與模式識別的硬體結構和程序設計。
  18. A device that converts analogue video and audio signals into digital for transmission over telecommunications facilities and also converts received signals back into analogue format

    一種將視頻和音頻的模擬信號轉換為數字信號進行傳輸，並將收到的數字信號轉換回模擬信號的儀器。
  19. Comparing with conventional nyquist converters, - converters greatly release the requirements for high performance of analog circuit and precisely matched components. additionally, these converters exploit the enhanced speed, circuit density and low cost of modern vlsi technologies. currentlly, - adcs have been widely used for audio a / d conversion

    - adc採用過采樣噪聲整形技術實現高精度模數轉換，和傳統的nyquist率模數轉換器相比，避免了對模擬電路性能指標和元器件匹配精度的較高要求，並可充分利用現代vlsi的高速、高集成度、低成本的優點，已成為音頻模數轉換的主要技術。
  20. Based on the analog vhf radios transmit / receive principals, chapter 4 illuminates simply the rf module, if digital module, and audio modules about the " zx08 " project, focused on the hardware design of the digital signal process chip " tms320vc5410 " and introduction of the audio module

    在介紹超短波模擬電臺收發原理基礎上，第四章對zx08項目中的射頻模塊、中頻數字化模塊、音頻模塊進行簡單的說明，重點是項目中的數字信號處理晶元tms320vc5410硬體設計和音頻部分的介紹。
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