音量壓縮 的英文怎麼說
中文拼音 [yīnliángyāsuō]
音量壓縮
英文
sound compression-
When we designed the system, we considered some key factors such as the system must deal with and store a great deal video and audio data, the capability of the hard disk in the server and client, hi the part of video compression, we choose the currently mpeg - 4 video constringent card to complete the collecting and compression of the video and audio signals
在論文研究與實用系統設計中,充分關注到大量視音頻數據的處理、存儲以及服務器和遠程客戶端的硬盤容量等關鍵因素,視音頻壓縮採用了最新的mpeg - 4視音頻壓縮卡進行視音頻信息的採集和壓縮。The focus is placed on the investigation of the standard of the encoding algorithm for mpeg audio layer iii, and the analysis of the major four modules in the compression algorithm, including encoding of subband filter bank, psychoacoustics model, quantification and huffman coding, frame packing
重點研究了mpeg音頻第層編碼的演算法標準。詳細分析了壓縮演算法中的四個主要功能模塊:子帶濾波器組編碼,心理聲學模型,比特流量化與霍夫曼編碼,幀數據流格式化。After about two years " insisting and hard working, this goal set at the beginning has become true. the developed c54x general assembly program for g. 729 speech signal compressing algorithm has passed the tracking with more than 3, 000 unitary standard measuring vectors. g. 729 speech signal compressing compiler using c54x general assembly program has been accomplished real - timely, and undistorted rebuilt speech signals have been obtained
因此本課題選用c54x的通用匯編語言編程實現g . 729語音壓縮編碼演算法,調試並通過了統一標準測試矢量三千多幀,最終在5402開發實驗板上實時實現了g . 729語音壓縮編碼器,獲得未失真的重建語音信號。The basic characteristics of the current data network are point - to - point, connectless, doing one ' s endeavor, no quality of service, etc. these characteristics do not meet the requirement of real - time services, therefore, the realization of voip need support of the some key technology. these technologies includes : speech sound coding and data compression, real - time transmission and control, mute compression and multicast, acoustic - echo cancellation and comfort noise generator, dynamic monitor and guarantee of quality of network service, as well as, the compatible of different network and different protocol with each other
但現有的數據網路的基本特性:點對點的、無連接的、盡力而為的、沒有服務質量保證等特性並不適合與實時的業務要求,因此voip的實現需要一些關鍵技術的支持,這些技術包括:語音編碼和壓縮技術、實時傳輸和控制技術、組播技術、靜音壓縮和舒適噪聲生成技術、回聲消除技術、網路服務質量的動態監測和保證技術、以及不同的網路、不同的協議之間的互連互通等等。The speech processing module can be divided into two parts, the first part includes speech compression, voice activity detection and echo cancellation module, which improve speech quality ; the other part includes dtmf and cpt module, which generate and detect some necessary telephony signal in the communication. this thesis is organized as follows
該語音處理模塊由兩部分組成,一方面是對語音的處理,包括語音壓縮模塊、靜音處理模塊和回聲消除模塊,主要為了提高voip的語音質量;另一方面是對電話通信的控制和處理,包括雙音多頻模塊和呼叫進程音模塊,主要為了產生和檢測ip電話通信中一些必須的電話信號。Our object is an intermediate frequency modem of a software defined radio transmitter - receiver of multi - service, multi - modulationmode and multi - processdatarate. first, related software defined radio theory is introduced ; later, channels of transmitter - receiver are designed with consideration of data format, modulation, fec, interwaving. and scrambling ; emphasis is placed on theory and implementation of an audio compression algorithm cvsd ( continuous variable slope delta modulation ) and a fec technique convolutional coding - decoding
本文首先介紹了相關的軟體無線電理論;然後完成了包括數據格式,調制方式,糾錯碼方式,交織器和擾碼器等部分的中頻數據機通道設計;接著著重介紹了系統中使用的音頻壓縮演算法cvsd (連續可變斜率增量調制)的原理和實現,以及作為前向糾錯碼的卷積碼編碼理論和編解碼的高效實現。Air conditioners, heat pumps and dehumidifiers with electrically driven compressors. measurement of airborne noise. determination of the sound power level
帶電動壓縮機的空調器熱力泵和除濕器.機載噪音的測量.噪音等級的測定To improve the speech quality, it is necessary to adopt such speech processing techniques like speech compression, voice activity detection, echo cancellation, jitter buffer and so on
為了提高語音質量,需要採取一系列的語音處理技術,主要包括語音壓縮技術、靜音處理技術、回聲消除技術、抖動緩沖技術等。This paper mainly discusses the design principles and chief techniques of a digital accessing system for power - line communication net ( plcn ). the technology of low bit rate speech compression high - speed modem based on plcn adaptive equalization to the channel anti - jamming and anti - fading are applied in this system. so speech tele - control data and tele - protection signals can be transmitted high quality in the band - limited channel
該系統綜合應用了低比特率語音信號壓縮編碼技術、基於電力通信網的高速調制解調技術、信號傳輸的通道自適應均衡技術和抗干擾、抗衰減技術,可在帶限通道中高質量的傳輸語音、遠動數據和遠方保護等信號,具有較高的整體性能。Vector quantization ( vq ) is an important technology in the field of image compression, which is widely used in various applications such as speech coding, audio and video compression, and teleconferencing systems
矢量量化( vq )是近年來圖像壓縮研究中的重要技術,廣泛應用於語音編碼、音視頻壓縮和遠程會議等系統中。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。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。National knowledge power bureau bureau chief king the evd standard accept when gather newsing reporter think, in the evd development process international rule of the right of knowledge of application of academic association of beginning of inside, local business enterprise is with the knowledge power strategy, become own core technique with standard, for the positive and international competition backlog of chinese business enterprise experience, grew with developped the own core competition ability to provide for chinese business enterprise good draw lessons from. according to all, the that new - released evd adoption have the right of independence knowledge compress the calculate way under the same code rate born and better than ratio ac 3 quantities s, come to a now the international and last compress the realm s advanced level
國家知識產權局局長王景川就evd標準接受記者采訪時認為,在evd的研發過程中,國內企業開始學會運用知識產權國際規則和知識產權戰略,形成自己的核心技術和標準,為中國企業積極參與國際競爭積累經驗,為中國企業培育和發展自己的核心競爭力提供了很好的借鑒。據悉,新推出的evd採用具有自主知識產權的音頻壓縮演算法eac在相同碼率下生成優于杜比ac 3質量的音頻,達到了目前國際上音頻壓縮領域的先進水平。Thus, owing to the rapidly growing of bandwidth on ip networks, internet - telephony that delivers a large amount of compressed voice ip packets to optimize the bandwidth utilization becomes one of the most favorite services
因此,隨著網路頻寬快速的成長,網際網路電話傳輸大量之壓縮語音封包儼然晉身熱門服務之一。For different applications, the audio part of mpeg - 1 provides three layers ; we select the implementation of the first layer on the ti tms320 dsp considering the complexity of the algorithm and the quality of voice. in first, the development and categories of speech coding and the standard of the mpeg series has been described
Mpeg - 1的音頻部分給出了三個層次以適應于不同的應用要求,綜合考慮演算法復雜度和話音質量要求,本課題選擇了mpeg - 1layer語音壓縮編碼方法並研究了其在titms320c6204dsp上的實現。Automatic volume compression
自動音量壓縮Thanks to recent technological advancements facilitating sound refinement and mp3 compression, the voice of a living master can now forever remain with us with better sound quality stored in higher capacities
拜現今科技之賜,結合電腦修音軟體和mp3資料壓縮等技術,以更佳的音質和高容量,將一位明師的法音永傳千古. .Methods of measuring the characteristics of reproducing equipment for digital audio compact discs ; iec 61096 : 1992 a1 : 1996 ; amendment a1 ; german version en 61096 : 1993 a1 : 1996
數字音響壓縮盤用重放設備特性的測量方法.修改a1Vorbis coding algorithm can provide high quality audio compression with advanced perceptual coding and transform coding technologies ; and because of its completely open and non - patent, vobis has attracted more and more attentions by music industrial and audio users
Vorbis編解碼演算法利用先進的感知編碼和變換編碼等技術提供高質量的音樂壓縮,同時因為其完全開源和無專利版權費用的特點而逐漸受到音樂工業和用戶的重視。The achievement of modifications on source - code was summarized as well. chapter one briefly introduced current developing status of audio coding techniques and the structure of this paper ; then chapter two shortly described the history of ogg vorbis and its technical process flow ; the data process and parameters calculations before vorbis quantization were discussed in chapter three ; and the details about vorbis quantization were shown in chapter four ; the following chapter five researched the process of vorbis decoding ; in chapter six, i derived and provided a unified implement structure on mdct and modified the source - code ; chapter seven is about some experiments where i compared and analyzed to finally present a summary on ogg vorbis encoding performance and the results on code modifications
論文第一章敘述了數字音頻壓縮技術發展狀況、音頻標準、主流音頻格式、各音頻格式存在的問題以及對本論文組織結構的簡要說明;第二章簡要介紹了oggvorbis音頻格式的概況和編解碼的技術結構;第三章詳細介紹了voibis演算法編碼過程中量化處理之前的數據處理和參數計算;第四章詳細介紹了voibis演算法編碼過程中的量化處理;第五章結合源代碼詳細介紹了vorbis演算法的解碼過程;第六章中對oggvorbis的mdct運算模塊提出一種統一實現方案,同時對代碼進行了修改;第七章將oggvorbis音頻與主流音頻格式mp3及aac進行了對比測試,對vorbis整體編碼性能和前面章節中代碼的修改作出了總結。分享友人