數字編碼語聲 的英文怎麼說

中文拼音 [shǔbiānshēng]
數字編碼語聲 英文
digit coded voice
  • : 數副詞(屢次) frequently; repeatedly
  • : Ⅰ名詞1 (文字) character; word 2 (字音) pronunciation 3 (字體) form of a written or printed ...
  • : Ⅰ動詞1 (編織) weave; plait; braid 2 (組織; 排列) make a list; arrange in a list; organize; gr...
  • : 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...
  • : 語動詞[書面語] (告訴) tell; inform
  • 數字 : 1. (表示數目的文字; 表示數目的符號) figure; digit; numeral; character; numeric character 2. (數量) quantity; amount
  • 編碼 : encoded; code; coded; encrypt; codogram; coding編碼表 encode table; 編碼程序 builder; 編碼尺 code...
  1. In this paper, combined with currently voice coding technique, espically with the fabulosity development of the mixed voice coding and the increasingly utility of the digital signals processor. we investigated the voice coding technique and discussed emphasizedly the technology of variable rate voice coding technology

    本文結合當前技術尤其是混合技術的驚人發展及信號處理器的日益實用化,研究了技術,並重點討論了變速率技術。論文簡要介紹技術中的波形器的主要性能。
  2. Once has the bandits and thieves to intrude guards against the place, the detector launches the wireless coded signal immediately, the networking center number which installs when is apart from defense area 150 meter within the main engine to send out the police whistle sound to report to the police immediately, reports to the police dials to establish in advance or reports to the police the telephone, the handset number, answers in the police telephone to return puts user pre - record to report to the police the pronunciation, long - distance reports to the police, simultaneously comes the real - time transmission through the internet to deploy troops for defense, to withdraw from a defended position, to report to the police and so on the condition, inquires the historic record through the computer network

    該系統還採用美國進口原裝晶元與先進的無線高頻技術微電腦cpu控制器主機組成。在防範地點安裝好主機后,並設置在布防狀態。一旦有盜賊闖入防範地點,探測器立刻發射無線信號,安裝在距防區150米以內的主機立即發出警笛報警,報警時撥打預先設定的聯網中心號或報警電話手機號,接警電話里回放用戶預錄的報警音,遠程報警,同時通過網際網路來實時傳遞布防撤防報警等狀態,通過電腦網路來查詢歷史記錄。
  3. These systems operate on a wide variety of continuous time signals, which include speech, medical imaging, sonar, radar, electronic warfare, instrumentation, consumer electronics and telecommunications systems ( terrestrial and satellite ). one of the key to the success of these systems has been advances in the development of the font end of the electronic systems - analog - to - digital converters ( a / d ) which converter the continuous - time stimuli signals to discrete - time binary - coded form

    這些系統可廣泛地應用於處理連續時間信號,包括音、醫學成像、、雷達、電子對戰、儀器、消費電器、遠程通訊(地面和衛星)等,而這些系統成功的關鍵因素之一就是電子系統的前端部件? ? a d取得了長足的進步( a d把連續時間信號轉換成離散時間、二進制信號,便於后級精確的信號處理) 。
  4. Both are used to separate the responsibility for rendering pages from the model and controller. both accept objects passed into them as an input argument, both allow inserting string values within code " expressions ", and allow direct use of java code to perform loops, declare variable, or perform logical flows " scriptlets ". both are good ways of representing the structure of a generated object web page, java class, or file while supporting customization of the details

    Jet與jsp非常類似:二者使用相同的法,實際上在後臺都被譯成java程序;二者都用來將呈現頁面與模型和控制器分離開來;二者都可以接受輸入的對象作為參,都可以在代中插入元串值(表達式) ,可以直接使用java代執行循環、明變量或執行邏輯流程式控制制(腳本) ;二者都可以很好地表示所生成對象的結構, ( web頁面、 java類或文件) ,而且可以支持用戶的詳細定製。
  5. We made an improvement in overcoming the defects in speech signal adaptive delta modulation ( abbr. adm ), such as slope overloading and grain noise. in this method, numerical sliding average filtering was used for filtering decoding speech signal. experiments and analyses indicate that the method makes waveforms in good agreement between the decoding of adm and the original pulse coding modulation ( abbr. pcm ) signal, and considerably improves, the playback speech quality in naturalness, legibility and under standability

    針對音信號自適應增量調制( adm )方式中斜率過載和顆粒噪缺點,提出了一種改進方法,它利用滑動平均方法對解后的信號進行濾波.試驗和分析表明,該方法使解后的信號波形與原脈沖調制( pcm )波形具有很好的一致性,使再生音質量在自然度、清晰度和可懂度方面比改進前均有較大提高
  6. At first, from the discrete digital model of the speech generation, the basic of speech encoding technology is briefly introduced in this thesis. secondly, some key techniques in the linear predictive speech coder of g. 723. 1 arithmetic are discussed in detail

    本文首先從音產生的離散模型出發,簡要敘述了的技術基礎,詳細討論了g . 723 . 1標準的線性預測器的一些關鍵技術。
  7. Speech enhancement is becoming an important branch of speech signal process, which mainly used in the noise reduction, the preprocess of the speech recognition system and lpc

    音增強目前已發展成為音信號處理的一個重要分支。它的主要應用是降低聽覺噪、作為識別系統的預處理和線性預測的預處理。
  8. It can satisfy the voip application demand at enterprise level and then performs an adjustment test according the vital factors bringing big affection to the quality of voice, such as voice quality and delay of different encoding styles, packaging capacity and delay, echo cacellation algorithm and digital voice mixture algorithm

    此網關能夠滿足企業級的volp應用要求。並在此網關設備上針對對音質量影響比較大的關鍵性因素進行了調整測試,如各類方式的話音質量與延遲、組包大小與延遲、回消除演算法、混音演算法等。
  9. After a great amount of detailed computer simulations and concise qualitative and quantitative theoretical analysis, the turbo codes " parameters and fpga specific hardware implementation architecture suitable for being integrated into dtv systems are determined. furthermore, the codec is completely designed with verilog hdl, ending with an occupation of less than a 600 - thousand - gate fpga chip. at this lowest hardware cost, a white noise snr threshold of 1. 8db at a net stream rate of 6mbps is achieved, which exceeds all other existent dtv systems " performance

    經過大量詳細的計算機軟體模擬和簡明扼要的定性與定量的理論分析,最終確定了電視系統中適合採用的turbo及針對fpga特殊構架的硬體實現結構,並用verilog硬體描述言完成了turbo器的完整設計,以佔用不到一片60萬門fpga晶元的較少的硬體資源取得了在6mbps凈率下1 . 8db的白噪信噪比門限這一遠遠超過現有任何電視系統的性能。
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