音頻帶寬 的英文怎麼說
中文拼音 [yīnbīndàikuān]
音頻帶寬
英文
ab audio bandwidth-
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。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
其中對語音編解碼器的設計採用優化g . 729a代碼達到設計要求，並在此基礎上加入g . 729b的靜音檢測模塊，以進一步降低網路傳輸帶寬;對回聲消除器的設計採用nlms演算法，通過設計自適應fir濾波器和語音檢測器達到回聲消除目的;對雙音多頻設計，信號發生端採用構造靜態參數表並通過二階正弦振蕩器產生信號，信號檢測端提取頻率信息以檢測信號;對呼叫進程音設計，除了類似雙音多頻的信號發生及頻率檢測設計外，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。Digital speech has preponderance over analog speech in reliability, robustness and security during communication. however, digital speech needs more bandwidth than the analog signal. especially with the requirement for communication frequency increasing, it ' s necessary to code speech signal at low rates
但是，數字化后的信號所佔的頻帶大幅增加，特別是在帶寬需求日益增長的今天，這個問題尤為突出，因此語音的低速率編碼（即壓縮編碼）成為迫切的要求。Relationship between the gyro ’ s bandwidth and sensitivity and the resonant frequency differential ratio is derived through frequency analysis. the bandwidth of the gyro increases as the resonant frequency differential ratio increases ; but the sensitivity decreases as the resonant frequency differential ratio increases
增大驅動軸和敏感軸之間諧振頻率的頻差，可以增加微陀螺的帶寬，但是降低了微陀螺的靈敏度，這為設計石英音叉結構參數時，確定驅動軸和敏感軸諧振頻率提供了指導依據。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。In the sonograms of psittacula agapornis after the section of right side nxiits the ascending parts of the tone disappeared. this causes their vocalizations became the descendent - call. and in the sonograms with the left side nxiits sectioned, the harmonic parts above the basic song were weaken or disappeared
牡丹鸚鵡在斷右側nxllts后其聲圖中音調上升支消失，因而其叫聲變為以降音調為主的降調叫聲；在斷左側nxllts后的聲圖中，其基本音以上的諧波成分減弱或消失而在斷雙側神經后其聲圖表現為一系列具有較低基頻的寬帶諧波譜。Speech communication is one of the most used modes in the digital trunking communication system. excellent algorithm of speech coding can save the bandwidth resource, improve the utilization of frequency, so it has important value for investigation
語音通信是數字集群通信系統中最常用的通信方式之一，優良的語音編解碼演算法能夠更加有效地節省帶寬資源，提高頻率利用率，因此具有重要的研究價值。The characteristic and key technologies of the system are as follows : ( 1 ) in realizing the live broadcast of audio and video, the problem of immense multimedia data and low networks bandwidth utilization ratio is solved by using mpeg - 4 as format of audio and video data. audio and video data are collected by video card cv500 which developed by beijing sum tone company ; meanwhile, the contradictory between the delay of networks transmitting and the quality of the image is well solved by setting a " bi - buffer area "
系統實現中解決的關鍵問題和特色主要有以下幾個方面： （ 1 ）在視音頻直播功能的實現中，通過使用北京算通公司的cv500視頻採集卡和cv500sdk進行視音頻數據採集，並採用當今最新的圖像和語音編碼壓縮標準mpeg - 4作為視音頻數據的採集格式，既保證了圖像的質量，又大大縮減了視音頻所佔的帶寬，從而解決了多媒體數據量大、網路帶寬利用率低的問題；同時，通過設置環形緩沖區的辦法來調和網路傳輸延時與圖像質量之間的矛盾，取得了較好的效果。Grading standard of quelity for broadband amplifier used in cabled distribution systems primarily intended for sound and television signals operating between 30mhz and 1ghz
30mhz 1ghz聲音和電視信號的電纜分配系統寬頻帶放大器質量分等標準Thus, the frequency domain adaptive filter is well suited to a multi - narrowband interference scenario, the paper studied frequency domain adaptive algorithms, carried out analysis and computer simulations. simulation results : for gold codes ds - ss signal of length 63, 4 interferences of signal to interference ratio ( sir ) 40 ub or 4 narrowband interferences of signal to interference ratio ( sir ) 40 db and frequency - spectrum bandwidth 12 percent of the whole bandwidth, the sir improvement is better than 20db
模擬結果指出，在輸入信號為干擾是4個等強度的多音干擾或4個頻譜帶寬占信號總帶寬12的強窄帶干擾（干信比為40db ）的擴頻信號（碼長為63的gold碼序列）情況下，演算法的干擾抑制比均優於20db 。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。Hd audio is also designed to prevent the occasional glitches or pops that other audio solutions can have by providing dedicated system bandwidth for critical audio functions
Hd音頻也是旨在防止偶然故障或持久性有機污染物的其他音頻解決方案可以通過提供專用系統帶寬為關鍵的音頻功能。The iboc technology, as proposed by the us - based ibiquity digital corporation, is also known as hd radio. it is designed to operate in the existing am and fm bands by attaching digital data to the sideband of analogue signals. the attached segment can either be a separate radio channel or other data
Iboc是美國ibiquity digital corporation所推廣的制式，又命名為hd radio ，它被設計成可以在現行的am和fm頻帶內運行，在模擬訊號的邊旁加上數碼訊號，它可以是電臺聲音廣播或是其它的數據，但由於頻寬的限制，最大約是200kbps系統設計雖然善用有限的頻譜，但技術上較為覆雜，目前正發展第二代產品。Almost any piece of information available at the time of interaction can be seen as context information : identity, spatial information ( e. g., location, orientation, speed and acceleration ), temporal information ( e. g., time of the day, date, and season of the year ), environmental information ( e. g., temperature, air quality, and light or noise level ), social situation ( e. g., who are you with, and people that are nearby ), resources that are nearby ( e. g., accessible devices, and hosts ), availability of resources ( e. g., battery, display, network, and bandwidth ), physiological measurements ( e. g., blood pressure, hart rate, respiration rate, muscle activity, and tone of voice ), activity ( e. g., talking, walking, and running ), schedules and agenda settings
幾乎任何在交互時可用的信息都能被看作環境信息：標識，空間信息（例如：位置，朝向，速度和加速度） ，時間信息（例如：某天的時間，日期，某年的季節） ，環境信息（例如：溫度，空氣質量，光或噪音的級別） ，附近的資源（例如：可訪問的設備，主機） ，可用的資源（例如：電池，顯示，網路和帶寬） ，生理度量（例如：血壓，心率，呼吸頻率，肌肉活動，語調） ，活動（例如：談話，行走，和奔跑） ，日程和內容設定。Realnetworks has formulated a specification for audio and video compression, known as realmedia. it is a multimedia application standard for the cross - platform clientserver structure, which prevails over the internet. it uses audiovideo streams and synchronous playback technologies to provide the best quality multimedia on a full broadband basis, and stereophony and continuous video at the transmission rate of 28. 8 kbps over the internet
Realnetworks公司所制定的音頻視頻壓縮規范稱為realmedia ，是目前在internet上相當流行的跨平臺的、客戶服務器結構的多媒體應用標準，它採用音頻視頻流和同步回放技術來實現在intranet上全帶寬地提供最優質的多媒體，同時也能夠在internet上以28Select whether audio or video should be emphasized in your surestream clips during degraded bandwidth conditions
選擇音頻或者視頻一定要降低你的源數據流的帶寬。With the continuing growth of the world wild web services over the internet, the demands of rapid image transmission over a network link of limited bandwidth and economical storage of a large image database are increasing rapidly
象聲音、圖像、視頻等這樣的多媒體信息往往佔用很大的傳輸帶寬和存儲空間。對它們的處理的工作量也往往非常大。Following the broadband service of audio and video to be provided, the recent method of narrowband access does not adapt to the requirement of the real - time service. so increasing the access speed will be emphasis to build information superhighways
隨著internet上話音、視頻等寬帶服務的發展，目前的窄帶接入方式已經不能滿足人們對實時性的要求，提升接入網的速度是形成信息高速公路的關鍵所在。This will use all available bandwidth after audio on the first channel for data
對第一通道的數據可用所有帶寬來模仿音頻。This task uses the ds - ss technology, and researchs the important technologies, such as synchronization of ds - ss, rake receive, narrow - band interference suppression, etc. these technology are useful for the voice - band channel of hf. i designed one system of very low bit rate for hf data communication, and completed the softwave module design of the system. i get the result of system ' s simulator, it ' s useful for the system realization
根據短波音頻帶寬通道的特點，重點對系統的擴頻同步技術、 rake接收技術和窄帶抗干擾技術進行了專題的研究。並根據自己多年的科研經驗，設計了一種具有實際意義的短波最低限度通信系統，並對實際的軟體實現進行了模塊化的設計。分享友人