音頻編碼器 的英文怎麼說
中文拼音 [yīnbīnbiānmǎqì]
音頻編碼器
英文
afc automatic frequency coder- 音 : 名詞1. (聲音) sound 2. (消息) news; tidings 3. [物理學] (音質) tone 4. (姓氏) a surname
- 頻 : Ⅰ形容詞(次數多) frequent Ⅱ副詞(屢次) frequently; repeatedly Ⅲ名詞1 [物理學] (物體每秒鐘振動...
- 編 : Ⅰ動詞1 (編織) weave; plait; braid 2 (組織; 排列) make a list; arrange in a list; organize; gr...
- 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
- 音頻 : [物理學] [電學] audio frequency; vf (voice frequency)音頻電路 voice frequency circuit; 音頻振蕩...
- 編碼器 : (將一項信息變換成一系列數碼信號的電路) coder; encoder; encipheror編碼器方框圖 encoder block diagram
- 編碼 : encoded; code; coded; encrypt; codogram; coding編碼表 encode table; 編碼程序 builder; 編碼尺 code...
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Code this control code of the recording s bit inside of the 16 the instruction to tell to put in the hdcd in the cd hdcd in the phonograph solution code, is central plains the number this of the high of that short restores out. like this, hdcd can in the cd of 44. 1 orotund number that khzs sampling frequency extreme limit inside, exceed the bandwidth 20 khzs re - appeared out
在hdcd編碼錄音的第16bit中的這個控制代碼指令告訴放在cd唱機中的hdcd解碼器,把訊號中原本的那個短促的陡高音還原出來。這樣, hdcd就能夠在cd的44 . 1khz的取樣頻率極限內,把頻寬超過20khz的聲音訊號重現出來了。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。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。Support for transcoding all supported audio formats using the converter component ( requires external commandline encoder executables for different output formats )
使用內建的轉換組件可以轉換所以支持的音頻格式(不同的輸出格式需要外部命令行編碼器可執行程序)With argumentation and comparing the difficulty of developing in different ways, in multimedia communication the terminal make up with mpeg - 4 encoder and decoder based on vw2010 chip, main controller based on x86 core and pc compatible stpc soc chip, ethernet adapter based on rtl8139. and this hardware platform can run linux operation system. the following works are done by auther. 1. participate in designing the project of multimedia terminal
經過論證和比較,權衡了開發難度和系統性能,在多媒體視頻會議終端設計中採用了基於x86內核片上系統晶元stpc的主控制器模塊,基於vw2010晶元的mpeg - 4音視頻編解碼器模塊,基於rtl8139晶元的10m / 100m以太網適配器模塊的硬體架構,主控制模塊運行linux操作系統,作為多媒體終端的軟體運行環境及軟體開發環境。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。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。Design of embedded audio system based on s3c2410 and uda
1341型立體聲音頻編解碼器的嵌入式音頻系統設計Specification for audio recording - pcm encoder decoder system
音頻記錄規范.脈沖編碼調制譯碼器解碼器系統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3We are developing advanced ip core of system lsi for digital av application, such as a video codec and an audio codec, with high performance, high quality, small size and low power consumption
我們正從事數字影音產品先進ip核的開發,例如高性能、高質量、小尺寸和低功耗的視頻編解碼器和音頻編解碼器。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。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。Audio codec requirements for the provision of bi - directional audio service over cable television networks using cable modem
使用電纜數據機通過有線電視網路提供雙向音頻服務的音頻編碼譯碼器要求Windows media encoder 9 series is a powerful tool for content producers who want to take advantage of the many innovations in windows media 9 series, including high - quality multichannel sound, high - definition video quality, new support for mixed - mode voice and music content, and more
Windows media編碼器9系列是一個強大的工具,對于希望充分利用windows media 9系列中的多項革新功能的內容提供商而言無疑是個福音,這些功能包括高質量多聲道音響高清晰度視頻質量新增對混和模式的聲音和音樂內容的支持以及其他更多功能。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。On the basis of analyzing the old system and theory, the element circuits of wireless digital audio transceiver modules are designed in detail including the digital audio encoding and decoding circuits with the surrounding circuits, the fsk circuit based on pll frequency synthesizer, the power amplifier circuit, the frequency discrimination and agc circuit
在分析原系統結構和理論的基礎上,完成了整個無線數字音頻傳輸模塊各單元電路的設計。主要包括有數字音頻編碼和解碼電路及外圍電路的設計、基於鎖相頻率合成器理論的fsk電路設計、功率放大器的設計、鑒頻與agc控制電路的設計。It includes the mepg2 demodulation module, the numeral regards the audio frequency coder and the conditional access module and so on
它主要包括mepg2解調模塊、數字視音頻編碼器和條件接收模塊等。Must contain mpeg4 encoder and can just make broadcasting and editor to mpeg4 file in the computer, the method is to install the plug - in package procedure, the software recommended is as follows, ffdshow, audio - visual storm stormcodec, klcodec204f, the hooligan s audio and video tool kit
電腦中必須裝有mpeg4編碼器才可以對mpeg4文件進行播放及編輯製作,方法是安裝插件程序,推薦的軟體有: ffdshow ,影音風暴stormcodec , klcodec204f ,阿飛的音頻視頻工具包Finally, on the basis of the mpeg - 1 layer hencoding hardware structure, the block of logic communicates with the pc over the parallel port and the interface for flash memory are design. then a mpeg audio coding system, which applies to store audio signal, is presented through the field programmable gate array device technology
最後,在mpeg - 1層編碼的硬體結構的基礎上,結合計算機並口通信和flash存儲器的介面模塊,採用現場可編程邏輯器件fpga技術,最終設計了一種應用於音頻信號存儲的mpeg音頻編碼系統。分享友人