語音帶寬 的英文怎麼說

中文拼音 [yǔyīndàikuān]
語音帶寬 英文
eech bandwidth
  • 語 : 語動詞[書面語] （告訴） tell; inform
  • 音 : 名詞1. （聲音） sound 2. （消息） news; tidings 3. [物理學] （音質） tone 4. （姓氏） a surname
  • 語音 : speech sounds; pronunciation; voice
  1. Digital speech technology has been widely used in many fields of communications in recent years, but it is just at beginning to use acoustical wave to implement underwater digital speech communications, which mainly results from the bandwidth - limited underwater acoustic communications channel and its high temporal and spatial variability

    目前，數字語音技術已廣泛應用於許多通信領域，但是在水下利用聲波進行數字語音通信卻是剛剛起步的新技術，這主要是由於水聲通道有限的通信帶寬及其時變、空變等特性所決定的。
  2. Escalators available in the lobby, 2 large function rooms, 8 break out rooms, 1 executive boardroom, large pre function area, latest audio visual equipment, simultaneous translation facilities, non - smoking rooms, hair and beauty salon, bistro atrium : 24 hour all day dinining international restaurant, four season : chinese cantonese and sichuan restaurant, robata - tei : japanese barbeque restaurant, rumours bar : cocktail bar with entertainment, le cafe : lobby lounge delicatessen and bar, 24 hour in - room dinining, undercover parking, gift and souvenir shop

    酒店擁有394間客房，包括30間套房，設施完善，功能備，每間客房均設有快速網際網路介面， 3部電話帶語音信箱，多用途電源插座，可移動的寬大寫字臺及舒適的辦公座椅，可連接個人電腦的數據介面。行政樓層提供特別的入住登記服務和小型會議室。商務中心設有獨立的工作間，配備電腦， 24小時免費網際網路介面及電子郵件服務，並提供文秘服務。
  3. 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。
  4. 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。
  5. Besides bandwidth and latency, what other parameter is needed to give a good characterization of the quality of service offered by a network used for digitized voice traffic

    數字化的語音所經過的網路，除了對帶寬和延遲有要求之外，還有什麼參數影響它的服務質量？
  6. Now we can transmit data, voice, and fax in communication lines whose bandwidths from 9. 6kb / s to 256kb / s through the voice / data multiplexing system. this method is good for improving bandwidth utilization and used widely in telecommunication, finance and military

    語音擻據復接設備可以實現在一條帶寬為9 . 6kb s 256kb s的用戶數字線路上實時傳輸數據、壓縮語音傳真的綜合業務，是利用帶寬的有效手段，在電力、電信、金融和軍事領域應用日益廣泛。
  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. 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。
  9. 311 well decorated bedrooms & amp; suites offering superior comfort and special amenities including dnn, satellite television, idd / ddd, mini - bar, voice ? cmail safe box and so on

    香山金源商旅中心酒店擁有各類客房311套。所有房間均提供寬帶上網埠、衛星電視、國際/國內直播電話、小酒吧、語音服務、保險箱等設施服務。現代化的設施、素雅的裝飾、使您倍感舒適方便。
  10. Speech coding systems are generally based on narrowband speech at present, but its frequency is restricted in 200hz ~ 3400hz and its sample rate is 8khz. with the development of the wideband speech, its bandwidth which is from 50hz to 7khz causes the quality of speech communication to approach in the feeling of face - to - face conversation, and makes the speech natural, expressive and comfortable. hence, it ’ s quite significant in researching on wideband speech coding system. in recent 20 years, the dsp and its software development kit has improved greatly, but the price has fallen sharply, thus it has more and more widespread applications now

    隨著當今世界的飛速發展，寬帶語音越來越受到人們的青睞，因為它的50hz 7khz的帶寬使得語音通訊質量接近於面對面交流的感覺，大大提高了語音的自然度、表現力和舒適度。因此，開發研製基於寬帶語音的編解碼系統具有十分重要的意義。在過去的短短二十年裡， dsp處理器的性能得到很大的改善，軟體和開發工具也得到相應的發展，價格卻大幅度地下降，從而得到越來越廣泛的應用。
  11. 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。
  12. The bandwidth is a principal factor that makes the speech quality bad. it is also the most difficult problem to be resolved for the internet

    帶寬是影響語音質量的一個主要因素，也是因特網當前最難解決的問題，一個解決的辦法就是採用高效的語音壓縮演算法。
  13. A voice system should ensure high quality voice first. meanwhile, the net bandwidth is very finite and precious resource

    語音通信系統首先必須保證良好的話音質量，而網路帶寬又是非常有限和寶貴的資源。
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    商務中心外幣兌換服務票務服務ddd電話idd電話理發美容室商場出租車全部房間有寬帶上網24小時客房服務商務中心精品商場國際長途電話泊車設施酒店保險箱服務客房送餐服務殘疾人設施旅行社及導游服務會議設施浦東機場班車美食店客房上網設施洗衣及乾衣服務外幣兌換櫃臺郵寄包裹服務語音留言信箱計程車及豪華轎車服務spa中心。
  15. Enhanced variable rate codec speech service option 3 for wideband spread spectrum digital systems

    寬帶頻譜擴展數字系統用增強的可變率編譯碼器語音服務選擇3
  16. With the rapid development of dsp technology, it has become a powerful tool in the wideband speech signal processing, which makes it possible to implement the complex algorithms for wideband speech in real time

    Dsp技術的高速發展為寬帶語音信號處理提供了強有力的工具，使得較為復雜的寬帶語音編解碼演算法的實時實現成為可能。
  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

    可選模式聲碼器（ smv ? selectablemodevocoder ）是3gpp2最新的用於寬帶擴頻cdma通信系統的變速率語音編碼標準，它實現了語音的多種低速編碼和解碼，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音的同時盡可能增加系統的容量，被認為是變速率語音編碼在cdma系統中應用的「突破性」技術，代表了當前語音編碼發展的方向和潮流，因此smv的研究具有很大的價值。
  18. In which adpcm is responsible for the audio signal ' s encoding / decoding and processing, and the latter is responsible for ip packing and unpacking

    前者用於語音的編解碼和信令提取，信令不單獨佔用帶寬；後者負責碼流的ip封裝和解封裝。
  19. 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。
  20. 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。
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