speech compression 中文意思是什麼

speech compression 解釋
亞音速壓縮
  • speech : n. 1. 言語；說話；談話；說話能力（或方式）。2. 民族語言，方言，專門語言；〈罕用語〉流言。3. 演說，演講；發言。4. 【語言學】詞（類）；引語；用語。5. （樂器的）音，音色。
  • compression : n. 1. 壓縮；壓緊；濃縮，緊縮。2. 加壓；壓抑。3. （表現的）簡練。4. 應壓試驗。
  1. The hardware of the ip phone codec to be designed is based on the fixed point digital signal processor ( ti ' s tms320vc5410 ) while the compression and decompression core in the software of dsp is based on the itu - t vg. 729a. ip phone codec carryout the task of collecting / playing - back. coding / decoding of speech signal and communication with embedded cpu. etc

    該語音編解碼器的硬體基於tms320vc5410 ，編解碼演算法遵循itu - tg . 729a協議，能夠實現語音信號的採集/回放、編碼/解碼以及同嵌入式cpu通信等功能，在8kbit / s的碼率下能夠提供獲得良好的語音質量。
  2. Experimental result shows that for sonant part of speech signal, 3 ~ 5 common ridges is enough to describe the main characteristics. signal compression is achieved by choosing proper way to represent the ridge information and use it to reconstruct the original signal

    在信號重建過程中，選擇合適的方法用少量數據來描述起關鍵作用的參數，並用這些參數來重建信號，可以達到信號壓縮的目的。
  3. 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。
  4. Embedded system - based ip phone speech compression technique

    電話高效語音壓縮技術
  5. At the receiving end, a inverse process is performed. the system receives low rate data and the fpga reorganizes a frame of data which is decoded by the compression chip every 20 ms. the obtained pcm signal is performed d / a to restore the analog speech signal

    在收端進行相反的過程，接收低碼率數據，並由fpga重新組幀，送至主晶元解碼得到pcm信號，再作d / a變換，恢復出模擬語音，系統是全雙工的。
  6. 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。
  7. With different modeling methods and quantization techniques, the speech compression schemes discussed in this thesis include : the compression based on general matching pursuit sinusoidal modeling, the compression based on sinusoidal modeling with perceptual gradient, the compression based on dynamic dictionary matching pursuit, the compression scheme using classified dynamic dictionaries, and the integrated compression scheme that combines the sinusoidal modeling with perceptual gradient and the classified dynamic dictionaries

    針對各種不同建模方法和參數量化技術，本文探討了基於普通匹配跟蹤正弦建模的壓縮編碼、感知梯度正弦建模壓縮編碼、基於動態字典匹配跟蹤的壓縮編碼、分類動態字典壓縮編碼，以及結合感知梯度正弦建模和分類動態字典的綜合編碼方案。
  8. 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。
  9. 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。
  10. So we must think out an effective way to settle them as quickly as possible, in this article, i put forward my scheme including image, speech, network and transmitting of file and so on to settle these problem. there are a lot of knowledge have been included in dealing with image ; they are image collection, image saving, image color distill ing, palette dealing with, image compression, image packing, image compounding, image improving etc. in dealing with speeh aspect, speech collection, speech compression, speech recording, speech playing have been involved.,

    在本方案中涉及到圖像、語音、網路、文件傳輸等方面的知識。其中的圖像部分涉及到圖像的採集、保存、顏色的提取、調色板的處理、壓縮、打包、合成、增強等方面的知識；語音方面涉及到語音的採集、保存、壓縮、錄取、回放等知識；網路方面涉及到基本的網路傳輸協議、組播、 winsock編程、網路阻塞事件的解決以及網路的監控等方面的知識；文件傳輸方面涉及到文件的分發、上傳、下載、刪除以及文件的傳輸協議ftp等知識。
  11. This paper accomplishes the hardware design and implementation of a speech compression system

    本文所完成的任務是用硬體設計並實現一個語音壓縮編/解碼系統。
  12. 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。
  13. Speech compression technologies are of great importance in increasing the capacities of ip telephone systems

    語音壓縮編碼技術對提高ip電話系統的系統容量有重要的意義。
  14. The hardware capacity for speech in spce061a is used and the function of speech compression is embedded in software

    該系統利用spce061a晶元具有的語音播放的硬體條件，並結合軟體演算法上的語音壓縮函數庫，實現了聲信號的智能化輸出。
  15. Recently, there is a great demand for video telephone, wireless communication and speech storage. speech compression is the key technology in communication areas

    現在，對可視電話、無線通信、衛星通信和語音存儲的需求越來越大，所以語音壓縮仍然是通信領域中的關鍵技術。
  16. With the rapid progress of digital communication technology and the increasing requirement in commercial application, speech compression technology has advanced rapidly in recent years

    近年來，隨著數字通信技術的發展及商業應用需求的增加，語音壓縮技術得到了迅猛的發展；本文所介紹的g
  17. Now, the standards of speech compression coding provide a way of transporting speech signals efficiently. in fact, all of them are to reduce the baud rate of data under definite speech quality

    相繼出現的語音信號壓縮標準為語音信號的高效傳輸提供了一種有效方法，其實質就是在相當的語音質量指標下，降低數字化語音的數碼率。
  18. H. 323 is an itu standard which provides several services including speech, data and multimedia etc. being a speech compression coder protocol supported by h. 323, g. 729 has the advantages of low bit - rate and high speech quality and been selected by itu - t as 8kb / s standard

    G . 729做為h . 323支持的語音壓縮編碼協議，具有低延遲，高語音質量的優點，被itu確定為8kb / s語音編碼標準。
  19. At first, the thesis briefly introduces the ip phone development status and standard, discuss its advantages and disadvantages, and then address the key technique issues of some commonly used speech compression standards

    本文首先介紹了ip電話的發展狀況和實現原理，分析了ip電話的特點和不足。接著論述了語音壓縮的關鍵技術，並對目前常用的一些音頻編解碼演算法作了簡要的介紹和比較。
  20. The system adopts an advanced specific speech compression chip based on ambe algorithm. it can achieve high compression rate at 4 kb / s. the system first performs a / d conversion on analog speech input to obtain pcm signal and the pcm signal was encoded by the main compression chip

    它採用了比較先進的基於ambe演算法的專用語音壓縮編/解碼晶元，壓縮率高，可以將數字語音信號碼率壓縮至4kb / s ，且編碼速率可通過修改控製程序選擇。
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