語音聲碼器 的英文怎麼說

中文拼音 [yǔyīnshēngmǎqì]
語音聲碼器 英文
phonetic vocoder
  • 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
  • 碼 : 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...
  • 器 : 名詞1. （器具） implement; utensil; ware 2. （器官） organ 3. （度量; 才能） capacity; talent 4. （姓氏） a surname
  • 語音 : speech sounds; pronunciation; voice
  1. On the basis of the study on the speech coder algorithms, paper describe an advanced method of developing dsp system software, and as the guidlines, we developed the programme of whole decoder unit. paper stress on analysis of the ecu in decoder unit. aiming at amr algorithms disadvantage of angularity of synthetical speech, paper study on the specutral extrapolation which apply to extrapolate reflect coefficient of track model to make error conceal processing of amr. at last paper analyze existing echo cancellation algorithms using on mobile communication system

    在此基礎上，描述了一種較為先進的大型dsp系統程序開發策略，並以此為指導思想，以美國ti公司c6000dsp開發平臺開發出了整個amr解碼器單元的系統程序。論文對amr解碼器的誤碼隱藏處理單元進行了重點分析，針對原有演算法合成語音自然度不好的缺點，論文研究了將譜外推法應用到amr演算法中外推出聲道模型反射系數參數進行誤碼消除處理。
  2. 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

    本文結合當前語音編碼技術尤其是混合編碼技術的驚人發展及數字信號處理器的日益實用化，研究了語音編碼技術，並重點討論了變速率語音編碼技術。論文簡要介紹語音編碼技術中的波形編碼和聲碼器的主要性能。
  3. The testing results show that the vocoder can synthesize high - quality speech when transmission rate is set to be 2. 4kbps

    非正式編解碼測試表明，本文的聲碼器能在2 . 4kbps的編碼速率上合成出較好的語音。
  4. 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。
  5. 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數字高頻技術微電腦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編碼信號，安裝在距防區150米以內的主機立即發出警笛聲報警，報警時撥打預先設定的聯網中心號碼或報警電話手機號碼，接警電話里回放用戶預錄的報警語音，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。
  6. The evrc vocoder is varialbe rate, the maximal encode rate is skbps, its voice quality is closed to qcelp - 13k, and has better ability of anti - disturbance

    Evrc聲碼器是可變速率的，最大編碼速率為8kbps ，在話音質量上接近於qcelp - 13k的語音編碼器，且具有更好的抗干擾能力。
  7. The demand is the power forcing speech coding to progress. traditionally linear prediction ( lpc ) vocoders are very efficient, which can encode speech from 800 to 2400bps, but unfortunately, artifacts such as buzzes, thump, and tonal noise always exist in them

    經典的線性預測（ lpc ）聲碼器具有很高的編碼效率，可以極低的碼率（ 800 2400bps ）對語音信號進行編碼，不幸的是它的合成語音聽起來很不自然，常常夾雜著嗡嗡聲，重擊聲或者音調噪聲。
  8. 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，包括語音、醫學成像、聲、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、二進制編碼的數字信號，便於后級精確的數字信號處理） 。
  9. Melp vocoders utilize mixed pulse and noise as the excitation to elimate the buzzes in traditional lpc vocoders, and add a jitter voicing state to overcome the tonal noise. parameters " interpolation, adaptive spectrum enhancement and pulse dispersion also are adopted to improve the continuity. the synthetic speech of melp vocoders sound much more natural and perceivable than the traditional vocoders "

    Melp聲碼器採用混合脈沖和噪聲激勵解決了經典lpc的嗡嗡聲的問題；引入了抖動濁音狀態以克服音調噪聲；利用參數插值、脈沖散布和自適應譜增強等措施提高合成語音的自然度和可懂度；此外還採用了多帶激勵，使其具有了比較強的抗背景噪聲的性能。
  10. Per channel, and the result from ti ' s software simulator in ccs ( code composer studio ) is given, putting forward the principles and difficulties of realization of this algorithm. the recovered music signal is very close to the original signal and they are difficult to tell apart. in this paper a scheme of real - time implementation for this algorithm is discussed

    本文敘述了mpeglayer音頻壓縮編解碼器的演算法及模擬實現研究，用c語言實現了的高保真音樂信號64kbps每聲道的非實時解碼，並在ti的ccs （ codecomposerstudio ）系統中的軟體模擬器上進行實時研究，提出了該演算法在具體實現中的要點和難點。
  11. Firstly, it introduces the development of speech coding, along with the significance of the low bit rate speech coding. it also compares the model of traditional dualistic excitation lpc vocoder and the multi - band excitation vocoder, and lucubrates the analytical method of frequency domain and time domain in the parameter extraction of multi - band excitation vocoding. secondly, based on the parameter extraction operation of keynote cycle, it adopts time domain in rough estimate operation of keynote and frequency domain in fine estimate operation of keynote, in according to the immediacy required in practice, to minish operation amount

    本文闡述了一種基於fpga的多帶激勵語音編碼器的研究與設計，首先介紹語音編碼研究的發展狀況以及低速率語音編碼研究的意義，接著對比分析了傳統二元激勵lpc聲碼器模型和多帶激勵編碼器模型，並深入研究了多帶激勵語音編碼參數提取的頻域和時域分析法，然後根據實際應用的實時性要求，為了減小運算量，在基音周期參數的提取的演算法實現上，本文採用在時域進行基音粗估運算，在頻域進行基音精細估計運算。
  12. In the next, we discuss the system of the meg - 1 layer i. the paper centers on the two kernel sub - parts : filtering coding and psychoacoustic model, do some research work in sub - band coding ( cbc ) theory and the relate theory such as quadrature mirror filter ( qmf ) and analyse sub - band filter ; also do research work in psychoacoustic theory especially the part related to the mpeg - 1 layer i. in the third chapter, introduce the ti tms320c6000 series dsps and their characteristics, also about the software development flow and the ti dsp / bios operating system of it. the forth chapter is the most important, firstly, according the algorithm flow in protocol, using c language validate the algorithm ; then, transplant and optimize the coding in dsp. in the processing of optimize, acording the assembler program characteristic of ti dsp, the paper put forward the analyse sub - band filter dsp optimization algorithm base on the eight spot idct. the algorithm has been optimize have greatly improved the work efficiency. make use of the technology of the dsp / bios host channels, data io pipe, software interrupt, we implement the musicam algorithm base on dsp / bios

    論文首先對當前語音編碼技術的發展、分類以及mpeg系列音頻標準作了介紹；接著在第二章，給出了layer的musicam （ masking - patternuniversalsubbandintegratedcodingandmultiplexing ）演算法的系統組成，圍繞分析子帶濾波器和心理聲學模型兩個核心模塊，深入研究了子帶編碼工作原理、比特分配及子帶編碼中用到的正交鏡像濾波器和分析子帶濾波器；探討了心理聲學基本原理和mpeg . 1layer所用到的心理聲學模型。第三章對titms320c6000系列dsp作了簡介，介紹了6000系列dsp結構特點、 c6000dsp軟體開發流程和tidsp / bios操作系統。第四章是本文的重點，首先根據協議給出的演算法用標準c語言編程實現並調試通過。
  13. G723 aerithemetic is a compressing arithemetic that proposed by itu - t and applied in speech and other audio frequency signals of low velocity multimedia services, such as : h. 323, h. 324 system. this arithemetic provides inspection to silence speech frames and fills in comfortable noise when it is silence. if optimize system and increase the complexity limitedly, we can get higher quality of speech. g723. 1 is also available in music or other voice signals, but the managing effect is not as good as speech ' s

    G . 723演算法是itu - t建議的應用於低速率多媒體服務中語音或其它音頻信號的壓縮演算法，例如： h . 323 ， h . 324系統。這種聲碼器具備兩種比特率： 5 . 3kbps ， 6 . 3kps 。在幀邊界處可以在兩種速率之間進行切換。
  14. The informal subjective test in real - time processing indicates that the synthetic speech quality of amr vocoder is better than it of rpe - ltp in gsm system, achieving toll quality, that can apply to devices employing the amr vocoder within the 3gpp system in terms of software and hardware

    實時處理的非正式主觀測試表明，合成語音質量優于gsm的rpe - ltp的語音質量，達到長途語音質量，完全可以實際應用，為第三代移動通信中聲碼器設備的研製奠定了良好的軟體和硬體基礎。
  15. 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。
  16. The paper makes great efforts on the software optimization of evrc vocoder. based on the understanding of tms320c64xx cpu structure, we do deeply - optimization on the loop which appear usually in voice signal processing, and this improve the utility ratio of cpu and the parallelity degree of cpu function cell. at the same time, we utilize the bit - exact test to test the fixed - point evrc vocoder with the test vectors of tia / eia / is - 718, which improve the robustness of the vocoder

    本文圍繞定點evrc聲碼器的軟體優化，做了大量的工作，在充分理解tms320c64xxcpu結構的基礎上，針對語音信號處理中大量出現的循環運算進行了深度優化，大大提高了cpu的利用率以及cpu功能單元的并行程度，同時，我們還用tia / eia / is - 718的測試向量對定點evrc聲碼器進行了嚴格比特對準測試，提高了聲碼器的魯棒性。
  17. Pitch detection is one of the most important tasks in low - rate speech coding field. the accuracy of pitch detection will affect the performance of the whole codec

    基音檢測是低速率語音編碼領域的一個非常重要的問題，準確檢測語音信號的基音周期非常關鍵，它直接影響到整個聲碼器的性能。
  18. 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

    本文首先從語音產生的離散數字模型出發，簡要敘述了語音編碼的技術基礎，詳細討論了g . 723 . 1標準的線性預測編碼聲碼器的一些關鍵技術。
  19. In traditional low bit - rate speech coding, considering that ears are not sensitive to phase information, the phase information is often neglected, and this will result in coarse and harsh speech quality, and it even may lead to inflection in pitch

    在傳統的低比特率語音編碼中，考慮到人耳對相位信息不敏感而經常忽略相位信息，這將導致語音粗糙、刺耳甚至音調發生改變。為了獲得高質量的聲碼器，語音的相位信息是不能不考慮的。
  20. At last this paper introduces a speech and data multiplexed based on multi - cpu and vlsi vocoders. the operation principle and characteristics of the multiplexed are analyzed

    介紹了採用多cpu系統和vlsi聲碼器為核心的語音數據復接器，分析了系統的工作原理和特點。
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