音量速率 的英文怎麼說
中文拼音 [yīnliángsùlǜ]
音量速率
英文
volume velocity-
The application of the software hardware advanced techniques, such as an algorithm for continuously outputting high - rate gps position data, voice data trucking system ( vdt ), a new gps ( avl ) system framework, an up - to - date mutual communication method and so on, enlarges the system capacity and the covering area, realizes the voice data transmitting in the present mobile communication channel, meets the need of kinetic positioning. in a word, it enhances the capability of management and decision
系統中gps定位數據的高速率連續輸出演算法、話音數據集群（ vdt ） 、新的gps車輛系統結構和全新的通信交互手段等先進技術的採用，人人提高了系統的容量和覆蓋面積，實現了在現有移動通信通道上數話兼容、高動態定位的需求，使科學管理和決策水平得到很大提高。In this thesis, the research focuses on pitch detection techniques of the low - rate wi speech coding. aimed at the problems of voiced - unvoiced error, pitch doubling and halving, accuracy of pitch detection and pitch quantization, a series of pitch detection techniques including pre - processing, pitch detection and pitch quantization were proposed
本文就低速率wi語音編碼中的基音檢測技術進行了深入研究，針對基音檢測中的清濁誤判、基音加倍減半、基音檢測精度及基音量化問題，提出了包括基音檢測前端處理、基音檢測演算法及基音量化的一整套基音檢測技術。It synthesizes the excellence of wave coding and parameter coding, adopts vector quantity, analyse - synthesize, perceptual weighting, therefore, gains good speech coding quality at 8kbit / s. cs - acelp can be used in individual telecom, iphone, c / n, microwave telecom and isdn
Cs - acelp演算法綜合了波形編碼和參數編碼的優點，以自適應預測編碼技術為基礎，採用了矢量量化、合成分析和感覺加權等技術，在8kbit / s速率上獲得了較高的語音編碼質量。At the present time, evrc is the best vocoder in the cdma system when take into account both the voice quality and the encode rate
在目前的cdma系統中，綜合語音質量和編碼速率， evrc是最佳的語音編碼器。Using ordinary modem, the data transmission speed under the current existing twisted pair access copper telephone lines network for voice communication is limited to 56kb s, which is totally inadequate for high volume data transmission. even using the adsl technology, the maximum transmission speed can only be increased up 6mb s. to surmount this problem by providing optical network access directly to all users ? home and offices not only involve very high cost but also extremely time consuming engineering work
現存原來用作話音通訊的雙絞銅線電話固網，在利用現時普通的modem情況下，數據傳送速度僅為56kb s即使利用較先進的adsl技術，其最高速率亦只有6mb s ，不足以應付高容量資訊的傳輸然而要做到光纖入戶，不但成本高昂，而且工程所需之時間漫長。Using ordinary modem, the data transmission speed under the current existing twisted pair access copper telephone lines network for voice communication is limited to 56kbs, which is totally inadequate for high volume data transmission. even using the adsl technology, the maximum transmission speed can only be increased up 6mbs. to surmount this problem by providing optical network access directly to all users ? home and offices not only involve very high cost but also extremely time consuming engineering work
現存原來用作話音通訊的雙絞銅線電話固網，在利用現時普通的modem情況下，數據傳送速度僅為56kbs ；即使利用較先進的adsl技術，其最高速率亦只有6mbs ，不足以應付高容量資訊的傳輸；然而要做到光纖入戶，不但成本高昂，而且工程所需之時間漫長。( 3 ) the most principal factors that influence the supersonic atomization process include the flow ratio of the gas - liquid metal ( gmr ) value, the flow of atomizing of gas and the range of the inverse vortex taper. the more of the value of three factors, the more advantage they are for the atomization and the more fine the powders are. ( 4 ) the produced powders are the best in efficient atomization efficiency, particle diameter, particle shape and dispersion when the solder alloy is zhl63a, atomizing medium is n2, the protrusion h = 6. 0mm, atomizing gas pressure p = 100mpa, over - heat temperature t = 167 ( t = 350 )
研究結果表明： （ 1 ）超音速霧化器的氣體流場在導液管下端形成一個倒渦流錐，在二維空間上呈軸對稱的雙峰分佈，負壓形成於這個倒渦流錐內； （ 2 ）修正後的霧化氣體速度公式可以滿足超音速霧化的要求； （ 3 ）影響超音速霧化工藝最根本的因素有氣液質量流率比（ gmr ）的大小、霧化氣體流量和倒渦流錐范圍，三個因素的值越大，對形成細粉越有利； （ 4 ）在焊錫合金為zhl63a ，霧化介質微n _ 2 ，導液管突出高度取h = 6 . 0mm ，霧化氣體壓力取p = 1 . 0mpa ，合金過熱度取t = 167 （ t = 350 ）時，所制得的粉末在有效霧化率、顆粒球形度、粒度及其離散度三個方面綜合性能最好。Subsequently, taking into consideration the characteristics of audio data over internet including delay, jitter, packet loss and etc., we propose a series of methods for solving this above problems, such as pre - storage technology, buffer technology, dynamic adjustment of the voice - coding rate to the state of network and integrated media synchronization playing mechanism, and etc. in the end, simulation on 10 / 100m lan is made using the above methods, and the result of the experiment demonstrates the method has good performance and can improve the quality of the audio data transmission
其次本文還深入研究了語音數據在非實時的internet數據網上的傳輸特性，這些特性包括延時、延時抖動、數據包丟失等。在本文的設計方案中提出了針對這些問題的解決方法，包括預取機制、設置緩沖區技術、動態速率調節技術以及媒體綜合同步播放機制等。最後採用這些方法在10 100m局域網上做了模擬實驗，實驗結果表明本文提出的方法是有效的，在網路狀況惡劣的情況下能夠改善語音播放質量。The quality of the existing low bit rate codecs is not satisfied
而目前已有的低速率編碼器，話音質量不夠理想。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。Fpga has become the best selection in the design of complex digital system in modern design process of digital system, especially in communication system, because of its merits like high integration, better reliability, short design period, less investment and agility. the usage efficiency of fpga for communication system can be raised by designing low rate speech coder in fpga
在現代數字系統設計中， fpga因為高集成度，高可靠性，設計周期短和投資小逐步成為復雜數字系統設計的理想首選，尤其是在通信系統中大量地使用，把低速率的語音編碼器在fpga中設計，可以提高通信系統中的fpga的利用率，節約成本。So this codec was optimized to represent speech with a high quality at the above rates using a limited amount of complexity
Tetra語音編解碼acelp演算法以低於4 . 8kbit s的編碼速率達到了令人相當滿意的語音質量。Among these achievements, the g. 723. 1 algorithm, which intergrates lots of advantages of other low bit rate speech coding algorithms, can help to get high voice synthesis quality at the rate of 5. 3kbit / s and 6. 3kbit / s
723 . 1是最重要的成果之一，它能以5 . 3kbit s和6 . 3kbit s兩種編碼速率提供較好的合成語音質量。By reducing coding rate, more speech signals can be transferred in the same channel. so, low bit rate speech coding has especially important significance when the transmission rate is limited very strictly
通過降低編碼速率，可以使同樣的通道容量能夠傳輸更多路的語音信號，在傳輸比特限制十分嚴格的場合，低速率語音編碼具有特別重要的意義。The valid approach to solve this problem is low rate, delay, loss and high quality codec
低速率、高質量和低成本的語音編解碼器成為解決這些問題的有效途徑。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。How to reduce the bit - rate of speech without remarkably degrading its perceptual quality is a question placed in the front of the researchers
如何在不犧牲語音通話質量的前提下盡可能降低話音信號傳輸的比特速率是擺在研究者面前的重要課題。As an important field, which influences the quality of ip phone, the quality of speech codec becomes the focus of research. the g. 729 is an 8kbit / s speech - coding standard issued by the international telecommunications union ( itu )
G . 729是國際電信聯盟（ itu ）頒布的編碼速率為8kbit s的低速率語音壓縮編碼標準，它採用了共軛結構算數碼本激勵線性預測（ cs - acelp ）技術，可以達到32kbit s的adpcm的語音質量。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。Many research institutes around the world are focusing on and developing wi speech coding algorithm, expecting to give a communication quality of synthesized speech at 2kb / s or even below
目前國際上許多研究機構正在集中研究和開發該演算法，期望在2kb / s或更低速率產生通信質量的重建語音。分享友人