語音壓縮器 的英文怎麼說
中文拼音 [yǔyīnyāsuōqì]
語音壓縮器
英文
vc voice compressor-
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。Ip voice gateway is a node between the communication control unit and the server of trunking system in trunking system network. its duty is digital compress coding and packetization of the voice in trunking system. it located on the edge of the network and accessed the trunking system into ip network from the kernel network angle
語音網關作為集群通信系統ip聯網的關鍵設備，它處于集群系統話路控制單元和集群系統服務器之間，起著語音壓縮、分組傳輸的重要作用;從網路的角度來看，它處于ip網路和集群無線通信系統之間，把集群通信系統接入ip網路。For the real time performance need of the low speed speech compress algorithm and the asic implement of the transfer process between programs, the design is put forward in the paper, in which state registers control the cross access between operator and memory, register windows are used for the parameters transfer, and the technique of hardware controlling is used to avoid pipeline conflict, so that the main problems of the transfer process in tr600 are solved effectively
摘要針對低速率語音壓縮演算法對處理器系統實時處理復雜運算的性能要求，就程序調用過程的asic實現問題進行了對比與分析，進而提出了用層次狀態寄存器控制存取運算元對存儲體交叉訪問的方法，並結合運用寄存器窗口傳遞參數的功能，以及利用空指令硬布線處理流水線沖突的方法，有效地解決了tr600晶元中調用過程存在的主要問題。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。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。With the great progresses and series of break - through in vlsi technologies, integrated dsp comes. integrated dsp has lots of advantages, such as low power cost, compactness, higher working speed and so on. dsp are very suitable in speech signal processing
隨著超大規模集成電路技術上取得的突破進展，高度集成化的dsp數字信號處理器具有體積小、功耗低和運算速度快等諸多優點，因此非常適用於語音信號的壓縮處理。This paper introduces a project of the wireless data transferring and the realization of speech encoding / decoding arithmetic based on the embedded system. in embedded system based on arm ? cpu, we accomplished the update of the system data by using the paging system, and emphatically researched how to avoid bit error. and, realizes the speech compression and decompression based on itu - t g. 729a, implement the speech synthesize of personal paging
在以arm7為處理器內核的嵌入式系統上，通過尋呼系統實現了系統數據的無線動態更新，重點解決了尋呼誤碼造成的數據錯誤等問題；以itu ? tg . 729a語音編解碼標準為基礎，通過語音壓縮與解壓演算法實現了個人尋呼的語音合成。The operating system and control program run on host processor mpc852t, the voice compress codec program run on the voice processor ac48304, they communication with the host port interface
主處理器mpc852t上運行操作系統及控製程序，語音壓縮分組處理器ac48304進行語音壓縮編解碼。In the paper, the embeded process platform and voice compression moudle were introduced firstly, then host port interface circuit was designed by which the host processor mpc852t communicated with ac48304. and then em interface and ethernet interface of the gateway were designed, which communicatewith trunking system and server respectively
本文的主要工作是在了解嵌入式處理平臺和語音壓縮模塊的基礎上，設計了powerpc處理器擴展ac48304的hpi介面電路;並設計了語音網關與集群系統的em介面以及以太網介面。In section two, we introduce some of the main coding theories that was in the g. 729, including the digital model of the production of speech signal, the linear prediction coding, vector quantization, lbg arithmetic and perceptual weighted filtering
第二章著重討論了g 729語音壓縮協議中所涉及到的語音編碼基礎理論。主要包括語音信號產生的數字模型、語音信號的線性預測分析、矢量量化及其lbg演算法和感知加權濾波器。The analoge voice from the trunking system was pcm coded, compressed coded, packtized and then transmitted to the server by udp / ip link in the voice gateway
語音網關把來自集群系統的模擬語音，經過pcm編碼，語音壓縮編碼，分組， rtp包封裝，經udp / ip鏈路傳輸到集群系統服務器。In the description of circuit design, the emphasis is paid the following hardware modules : ad / da inverter, dsp module, external program / data memory, cpld control logic, serial communication module, power module, and so on. problems and the corresponding solutions found in the design and debug stage are discussed, too. finally, the low - level software driver design is presented in detail, including system booting, initialization of dsp registers, cpld logic and timing control, drivers for asynchronous communication fifo, and drivers for ad converter
在電路模塊分析中，重點介紹了語音的輸入放大和輸出緩沖部分、 ad da轉換、 dsp語音壓縮解壓、外部程序數據存儲器、 cpld邏輯控制、串列收發組件、電源供電以及dsp的jtag介面等等，並且給出了在硬體電路設計和調試過程中的問題與解決辦法。It is configured to support conversions from html to wml, html to chtml compact html for i - mode phones in japan, and html to vxml voice xml for voice servers
它被配置成支持從html到wml 、從html到chtml （用於日本i - mode電話的壓縮格式的html ）以及從html到vxml （用於語音服務器的語音xml ）的轉換。分享友人